Asterisk Information: 2007

Monday, December 31, 2007

Re: [asterisk-users] One Way Delay in Audio Over Analog

Brian Alexander wrote:
> I have been trying to track down the cause/fix for a problem and I am
> out of ideas... I am hoping one of you can point me in the right direction.
>
> The symptom is that when a calls is placed from an internal extension
> through an analog line to a number on the pstn the caller can hear the
> callee but the callee can not hear the caller for as long as ten seconds.
>
> The problem appears to happen fairly consistently on the same pstn
> numbers. However, I have not seen a common characteristic in those
> numbers. For example, one of them is a direct number to a cell phone and
> another is to a Verizon fiber-optic phone/data service.
>
> The problem does not seem to be related to the type of SIP phone being
> used by the caller - for example, we have tried both X-Lite and Polycom
> phones without a change in behavior.
>
> The problem does not appear to occur if the callee then calls into our
> system (at least the one time I was able to have this happen).
>
> Turning on or off echo cancellation and/or call progress does not seem
> to change the behavior.
>
> I will appreciate any ideas you have. I am certainly stumped.
>
> Thanks and Happy New Year!
> -Brian

Brian,

What about some facts ?

Hardware ?

Software versions ?

/Mats

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Re: [asterisk-users] Asterisk 1.4 Fax

Rob Hillis wrote:
> Last time I heard IAXModem didn't support T.38 because the IAX2
> protocol didn't support T.38 - whether that's still the case or not, I
> don't know.
There are actually two reasons. One is that T.38 over IAX is not
defined. The other is the current T.38 termination support in spandsp is
only for the full FAX machine it contains. T.38 termination to the class
1 FAX modem (T.31) interface for HylaFAX is a work in progress. When
that is done, I hope we will have a sipmodem to replace iaxmodem,
offering bother audio and T.38 to HylaFAX functionality.

Steve

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Re: [asterisk-users] Asterisk 1.4 Fax

If by "fax box" you mean an ATA with a fax machine attached them
Asterisk 1.4 with T38 passthrough should work if the SIP provider has
T.38 capabilites.

If by "fax box" you mean a 'faxmail inbox' then no Asterisk cannot
help you terminate that from SIP. Get a Cisco gateway, make sure your
provider uses T.38 and connect that to your Asterisk via T1 or E1.

On Jan 1, 2008 12:50 AM, Al lists <asteriskal@gmail.com> wrote:
> at this time is terminating a SIP trunk,
> each DID will get its own fax box.
> I guess at this time i'm looking to find a tutorial for installing iaxmodem
> and hylafax as it seems to be the answer.
>
>
>
>
> On Dec 31, 2007 9:11 PM, Andrew Joakimsen <joakimsen@gmail.com> wrote:
> >
> >
> >
> >
> > On Dec 28, 2007 8:28 PM, Al lists <asteriskal@gmail.com> wrote:
> > > what method is preferred:
> > > haylafax and Iaxmodem or spnadsp for faxing.
> > >
> >
> > What are you trying to do and do you have a T1 or ISDN line?
> >
> >
> >
> >
> >
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> >
>
>
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Re: [asterisk-users] Asterisk 1.4 Fax

Unless your provider provides a T.38 gateway, fax over SIP is pretty much guaranteed to be unusable. Often you can get away with it over a LAN using G711a or G711u, but any of the lower bandwidth codecs won't be able to properly handle fax calls.

Whilst I haven't used it myself, I believe IAXmodem and Hylafax are used for sending and receiving faxes from a local PSTN termination point such as T1 or ISDN.

The IAXmodem web site explains the pitfalls of faxing over the internet. See http://iaxmodem.sourceforge.net/faq.php for more info. Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know.

Al lists wrote:

at this time is terminating a SIP trunk,
each DID will get its own fax box.
I guess at this time i'm looking to find a tutorial for installing iaxmodem and hylafax as it seems to be the answer.

On Dec 31, 2007 9:11 PM, Andrew Joakimsen <joakimsen@gmail.com> wrote:

On Dec 28, 2007 8:28 PM, Al lists <asteriskal@gmail.com> wrote:
> what method is preferred:
> haylafax and Iaxmodem or spnadsp for faxing.
>

What are you trying to do and do you have a T1 or ISDN line?

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Re: [asterisk-users] Asterisk 1.4 Fax

at this time is terminating a SIP trunk,
each DID will get its own fax box.
I guess at this time i'm looking to find a tutorial for installing iaxmodem and hylafax as it seems to be the answer.

On Dec 31, 2007 9:11 PM, Andrew Joakimsen <joakimsen@gmail.com> wrote:

On Dec 28, 2007 8:28 PM, Al lists <asteriskal@gmail.com> wrote:
> what method is preferred:
> haylafax and Iaxmodem or spnadsp for faxing.
>

What are you trying to do and do you have a T1 or ISDN line?

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Re: [asterisk-users] Asterisk 1.4 Fax

On Dec 28, 2007 8:28 PM, Al lists <asteriskal@gmail.com> wrote:
> what method is preferred:
> haylafax and Iaxmodem or spnadsp for faxing.
>

What are you trying to do and do you have a T1 or ISDN line?

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Re: [asterisk-users] Polycom Digit Map

Doug wrote:
> At 14:27 12/31/2007, Mojo with Horan & Company, LLC wrote:
> >Mojo with Horan & Company, LLC wrote:
> >> So try: 011XXXXXXXXXXT in your digit map, meaning "011 plus at least six
> >> digits, consider it good"
> >Err duh, that's ten X's not six :) To account for the Tajikistan
> >example plus a little bit of local number.
> >
> >Really, it's dead simple to just do it like "011XT", which means 011
> >plus ANYTHING else plus a timeout :)
> >
> >Moj
>
> I think you might need a dot "." in there to
> accept any length:
>
> dialplan.1.digitmap="*xxx|*xxxx|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|xxxT"
> dialplan.1.digitmap.timeOut="3"
>
Oooh, too true. Thanks for remembering!

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Re: [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone

glenn,
check your handset cord... it might be plugged into the wrong port in the back of the phone. perhaps the headset jack...
daveC

Glenn Gillen wrote:

Hey all,  I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it:  - Registers correctly - Is able to make calls to other peers  However it is not able to answer calls made to it. That is, the handset actually rings, but I've no way to answer it. The answer soft key, picking up the phone, etc. all have no effect. And I'm at a loss as to what setting should be altered to fix it. Any ideas?  Possibly a tangent, but also affecting this handset, is that trying to dial out over an external SIP trunk fails on the first attempt. But calling an internal peer and then trying a second time makes it mysteriously work.  Any help greatly appreciated,

--  My wife's sister is in California.   I should buy her a Videophone2008!  Truly, The Next Best Thing to Being There! --  WorldWideVideoPhones.com 856.380.0894

[asterisk-users] One Way Delay in Audio Over Analog

I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction.

The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds.

The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service.

The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior.

The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen).

Turning on or off echo cancellation and/or call progress does not seem to change the behavior.

I will appreciate any ideas you have. I am certainly stumped.

Thanks and Happy New Year!
-Brian

Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

On Sunday 30 December 2007 14:40:40 Mindaugas Kezys wrote:
> Thank you!
>
> Will it come to 1.4.16.3 or 1.4.17?

Yes, it will.

--
Tilghman

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Re: [asterisk-users] Polycom Digit Map

At 14:27 12/31/2007, Mojo with Horan & Company, LLC wrote:
>Mojo with Horan & Company, LLC wrote:
>> So try: 011XXXXXXXXXXT in your digit map, meaning "011 plus at least six
>> digits, consider it good"
>Err duh, that's ten X's not six :) To account for the Tajikistan
>example plus a little bit of local number.
>
>Really, it's dead simple to just do it like "011XT", which means 011
>plus ANYTHING else plus a timeout :)
>
>Moj

I think you might need a dot "." in there to
accept any length:

>
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Re: [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone

On Mon, 2007-12-31 at 21:13 +0000, Glenn Gillen wrote:
> I'm having some difficulty getting the Polycom hardphone to function
> correctly. Watching the logs and debug trace it:
>
> - Registers correctly
> - Is able to make calls to other peers
>
> However it is not able to answer calls made to it.

The first thing I'd do would be to capture a SIP trace of the call,
using either "sip set debug" from the Asterisk CLI, or a packet sniffer
such as tcpdump or Wireshark. I'd also turn up the core verbosity at
the Asterisk CLI and look for clues that might be present there while
the call is being made.

---
Jared Smith
Community Relations Manager
Digium, Inc.

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[asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone

Hey all,

I've setup my asterisk install on a CentOS5 server, I've got a few
IAX2 and SIP softphone clients connected on the same subnet and at
least 1 external IAX2 softphone. However I'm having some difficulty
getting the Polycom hardphone to function correctly. Watching the logs
and debug trace it:

- Registers correctly
- Is able to make calls to other peers

However it is not able to answer calls made to it. That is, the
handset actually rings, but I've no way to answer it. The answer soft
key, picking up the phone, etc. all have no effect. And I'm at a loss
as to what setting should be altered to fix it. Any ideas?

Possibly a tangent, but also affecting this handset, is that trying to
dial out over an external SIP trunk fails on the first attempt. But
calling an internal peer and then trying a second time makes it
mysteriously work.

Any help greatly appreciated,

--
Glenn

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

On Mon, 2007-12-31 at 14:16 -0600, Kevin P. Fleming wrote:
> No. The files to repopulate the CF card are available to users who have
> active support subscriptions and they can replace the card. Users can
> also, of course, make a backup copy of the card on a new card when they
> receive the unit and have a ready-to-install replacement should any
> problems occur.

That's all fair enough then. I was just concerned with the message that
was being sent along with the "replacing the CF card is unsupported"
message.

Re: [asterisk-users] Polycom Digit Map

Mojo with Horan & Company, LLC wrote:
> So try: 011XXXXXXXXXXT in your digit map, meaning "011 plus at least six
> digits, consider it good"
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a little bit of local number.

Really, it's dead simple to just do it like "011XT", which means 011
plus ANYTHING else plus a timeout :)

Moj

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

Kevin P. Fleming wrote:

> the Linux kernel on the AA50 does not have NFS
> support nor SMB support, and there are no userspace tools present to
> handle NFS or SMB mounting of filesystems.

FUSE? But it's probably not on the appliance.

Regards,
Philipp Kempgen

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

Brian J. Murrell wrote:

> And how long will that flash card last with the log and vmail churn?
> Flash devices have a limited number of writes you can do to a single
> cell before it "wears out" and cannot be written to any more.

All modern flash cards (not flash chips, which are lower level) have
built-in wear leveling. There is still an upper limit to what the card
can handle, but keeping in mind the target market for this device (a
small office with less than 50 users) it's not likely that the voicemail
volume is going to be so extreme as to wear out the CF card. We do not
ship the AA50 with Asterisk logs enabled to the CF card as far as I
know, primarily for this reason.

> So does one have to throw out the whole appliance when one wears out the
> flash card?

No. The files to repopulate the CF card are available to users who have
active support subscriptions and they can replace the card. Users can
also, of course, make a backup copy of the card on a new card when they
receive the unit and have a ready-to-install replacement should any
problems occur.

--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

Gregory Malsack wrote:

> ------------------ Digium ------------------
>
> That's news to me as well as the rest of the sales team. We were told that users cannot change the 1gb flash card.
>
> I just spoke with one of the Sales Engineers and he stated that it is apparently possible to change out the flash card on the AA50. However, it is not supported because the read/write speeds could be different on a new flash card, and thus not work.

That is correct; we would not recommend using just *any* CF card, as the
write speed of the card needs to be pretty high to be able support
multiple voicemail messages being written simultaneously. With that
said, though, it is possible to use a higher capacity CF card, but my
previous response that said it was 'easy' was a bit of an overstatement
:-) It can be done, and our support department does know how to get you
the files you would need to populate the replacement card.

> ------------------ Digium ------------------
>
> It is not possible to do so. However, the 1gb Flash card on the appliance will store up to 3000 minutes worth of voicemail.

This is correct as well; the Linux kernel on the AA50 does not have NFS
support nor SMB support, and there are no userspace tools present to
handle NFS or SMB mounting of filesystems.

--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Polycom Digit Map

Doug Lytle wrote:
> Michael Munger wrote:
>
>> only connects me to a dial tone and says "Enter More Digits."
>>
>>
>
> It actually says this?
>
> I would say then it's not the phone, but your phone system's
> programming. The Polycoms don't verbally say anything, at least not the
> ones I deal with.
>
> Doug
>
>
>
No it doesn't SAY it -- the polycoms put on the screen "Enter more
digits". I think it's when what you've dialed doesn't match an entry in
your digit map, or possibly when asterisk says that extension does not
match anything....

So try: 011XXXXXXXXXXT in your digit map, meaning "011 plus at least six
digits, consider it good" because you can't know how long the string
will be in advance. You want to allow for the smallest possible, which
I suspect would be a three digit country code, like in Tonga (676) --
and you want to allow for the longest possible, to account for stuff
like in Tajikistan: 992 37962 is BEFORE the local number, so you'd want
011+ at least 9 Xs following it 011XXXXXXXXXX -- Tricky!

**
Moj

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Re: [asterisk-users] Polycom Digit Map

Michael Munger wrote:
> only connects me to a dial tone and says "Enter More Digits."
>

It actually says this?

I would say then it's not the phone, but your phone system's
programming. The Polycoms don't verbally say anything, at least not the
ones I deal with.

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

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Re: [asterisk-users] Realtime & sip.conf

I don't understand the USERS vs PEER vs FRIENDS. I just use Peer for everything. Has to do with "can I only contact you or can you contact me too?" ... Peer does it all.

RealTime does have an issue. If you don't turn on caching, then it holds no state information. So if you think you're going to encouter firewall issues and need NAT=yes, then realtime will run in a static mode where you'll need to reload each time you change anything (like a password). I think the proper command is something like "SIP PRUNE".

Finally, putting something like sip.conf into realtime wasn't a move I wanted to make. I simply generate a SIP.conf file myself via my own program and run a SIP RELOAD (or simply reboot) each time I make a big change. Changes don't happen often so no biggie, where as I did want to make live changes to other SIP users without reloading (like a person using our web interface to change their own password).

On 12/29/07, hugolivude <hugolivude@gmail.com> wrote:

Hi -

I'm looking into realtime and I'm having a bit of a problem with the SIP part.

My review of the posts seems to indicate that I should use realtime
static for the [general] part of my sip.conf including the
registration commands:

   register=><did>:<secret>@<domain>/<did context>

and use realtime realtime (funny name!) for peers and friends:

[myprovider]
type=peer
auth=md5
username=...
fromuser=...
fromdomain=...
secret=...
host=...
port=5060
nat=yes
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=incoming-sip

Is this correct?  What's throwing me off is this statment found @
http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static :

   NOTE: You can only store a static config OR a RealTime config. You
cannot, for example, store
              sip.conf and use sipfriends via RealTime.

If I am correct, it would suggest that I'll have to do a reload when I
add a DiD, but a reload won't be necessary if a new SIP client is
added.  Do I have it right?

Also, what's the difference between a peer and a user?  I used to
think that a "user" was an agent  authorized to call in to my * box, a
"peer" was an agent I could reach and a "freind" was both.  What's
throwing me off now is the statement found @
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14966:

    With newer versions of Asterisk the concept of SIP 'users' will be
phased out.

I can't understand this especially in the context of extconfig.conf
that uses both a sipuser and sippeer entry.  Could someone clarify for
me?

Thanks,
H

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--
/Nick

Re: [asterisk-users] Polycom Digit Map

That was one of the many iterations I tried already. It seems to respond
in that it recognizes that I am dialing 01186106887XXXX, but then it
only connects me to a dial tone and says "Enter More Digits."

There has to be something simple I am over looking here. I understand
regular expressions, etc... I do have a tendancy to make a problem more
complex than it really is though!

Yours,

Michael Munger, dCAP
404-438-2128
michael@highpoweredhelp.com

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jerry
Jones
Sent: Monday, December 31, 2007 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Digit Map

On Dec 31, 2007, at 11:36 AM, Michael Munger wrote:

> I need the digit map to call China. Example number:
>
>
>
> 011-86-10-6887-XXXX
>
>
>
> 011-International (obvious)
>
> 86 is country code (China)
>
> 10 is city code (Beijing)
>
> Last 8 digits are the number.
>
>
>
> I tried using 011xxx.T but it always asks me to enter more digits.
> Tried some variations as well, but no joy.
>
>
>
Yours should work if you wait long enough for t to timeout.

How about 01186xxxxxxxxxx?

Plus, IARC, when dialing offhook, pressing # should terminate dialing
and send what it has at that point.

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

Brian J. Murrell wrote:
>
> And how long will that flash card last with the log and vmail churn?
> Flash devices have a limited number of writes you can do to a single
> cell before it "wears out" and cannot be written to any more.
>
> So does one have to throw out the whole appliance when one wears out the flash card?
Well it IS an appliance, after all.
And that IS the American way! Throw it out when done with it.
We do it with appliances, wives and our pets!

John Novack

--
Dog is my co-pilot

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Re: [asterisk-users] app_echo.c

I would GUESS that if this line is removed, asterisk is settling on slin
codec for the channel and does not try to negotiate anything better?
Hence it will work without it.

Mojo

Bhrugu Mehta wrote:
> hi, all
> I have test echo application for just fun.
> I can'nt understand why this is used below in .c file,
>
> format = ast_best_codec(chan->nativeformats);
> ast_set_write_format(chan, format);
> ast_set_read_format(chan, format);
>
> without this this application work fine.
> then why this is used.
>
> any suggestion??
>
> Bhrugu mehta
>
> _______________________________________________
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>

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>

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Re: [asterisk-users] Polycom Digit Map

Jerry Jones wrote:
> Yours should work if you wait long enough for t to timeout.
I think your digit map needs a T on the end of it if you want to allow
timeouts for that match.

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

On Mon, 2007-12-31 at 12:02 -0600, Gregory Malsack wrote:
> Here is some information I received from my account rep at Digium regarding this information:
>
> ------------------ Digium ------------------
>
> That's news to me as well as the rest of the sales team. We were told that users cannot change the 1gb flash card.
>
> I just spoke with one of the Sales Engineers and he stated that it is apparently possible to change out the flash card on the AA50. However, it is not supported because the read/write speeds could be different on a new flash card, and thus not work.
>
> ------------------
>
> Also here is what I was told when I asked my digium rep if we could nfs or samba mount additional storage space to the appliance (since it does run on linux).
>
> ------------------ Digium ------------------
>
> It is not possible to do so. However, the 1gb Flash card on the appliance will store up to 3000 minutes worth of voicemail.
>
> ------------------

And how long will that flash card last with the log and vmail churn?
Flash devices have a limited number of writes you can do to a single
cell before it "wears out" and cannot be written to any more.

So does one have to throw out the whole appliance when one wears out the
flash card?

Re: [asterisk-users] Directories Used by Asterisk

It is when you type 'make install' that these directories get created.
'make linux26' IS obsolete as another poster mentioned.
broadband Voice wrote:
> I successfully obtained the Asterisk code and extracted them into
> /usr/src. When I make and install asterisk, zaptel, libpri etc. Are
> they supposed to move automatically into their respective directories?
>
> I cannot find:
>
>
> /etc/asterisk/
>
> /usr/lib/asterisk/modules/
>
> /var/lib/asterisk
>
>
>
> Do I have to manually create them or this is failed install? Thanks.
>
> ------------------------------------------------------------------------
>
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>

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Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

Here is some information I received from my account rep at Digium regarding this information:

------------------ Digium ------------------

That's news to me as well as the rest of the sales team. We were told that users cannot change the 1gb flash card.

I just spoke with one of the Sales Engineers and he stated that it is apparently possible to change out the flash card on the AA50. However, it is not supported because the read/write speeds could be different on a new flash card, and thus not work.

------------------

Also here is what I was told when I asked my digium rep if we could nfs or samba mount additional storage space to the appliance (since it does run on linux).

------------------ Digium ------------------

It is not possible to do so. However, the 1gb Flash card on the appliance will store up to 3000 minutes worth of voicemail.

------------------

Sincerely,
Gregory Malsack
President
Classic Services
Select Digium Reseller

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Saturday, December 29, 2007 7:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium Asterisk Appliance voicemail & logs

Barry D. Hassler wrote:
> Does anyone know how much space the appliance has for voicemail and/or
> logs? Doesn't have an embedded disk from what I can see, and only a 1G
> flash card?

Correct. Nearly all of the 1GB CompactFlash card is available for
voicemail, logs, CDRs, etc, and of course larger CompactFlash cards can
easily be used.

--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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No virus found in this incoming message.
Checked by AVG Free Edition.
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[asterisk-users] How to use AddQueueMember with IAX2 peers?

Hi all,

I've been working on this for days and can't find a solution. I need to use
AddQueueMember for my agent logins to my Queues -- but a number of my agents
are outside the main server, which is connected to my asterisk network over
IAX2. I can't just do a AddQueueMember(queuename) because it puts in a
complicated member calleridnum like: IAX2/peername:65723/23
Which won't exist when it comes time to transfer a call to that member.

Help!

I have tried using the chan_local formatted strings instead like
Local/calleridnum@iax2context -- but you lose all sorts of functionality if
you do it that way

--
--
Chris

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Re: [asterisk-users] Polycom Digit Map

On Dec 31, 2007, at 11:36 AM, Michael Munger wrote:

How about 01186xxxxxxxxxx?

Plus, IARC, when dialing offhook, pressing # should terminate dialing
and send what it has at that point.

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[asterisk-users] Polycom Digit Map

I need the digit map to call China. Example number:

011-86-10-6887-XXXX

011-International (obvious)

86 is country code (China)

10 is city code (Beijing)

Last 8 digits are the number.

I tried using 011xxx.T but it always asks me to enter more digits. Tried some variations as well, but no joy.

-Michael

Re: [asterisk-users] IVR help, please

Doug Lytle wrote:
> Jay Moore wrote:
>> Hi list.
>>
>> I'm new to IVRs and trying to set up one that toggles an auto-forward
>> flag on or off for specific accounts.
>>
>>
>
> Why don't you post what you've currently written and we'll go from there?
>
> Doug
>

Actually, after switching to AEL, I think I finally got it working
properly. Thank you for your response, however.

Jay

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[asterisk-users] Require IP Phones in Pakistan

Hello All,

We need IP Phones in Lahore, Pakistan. Preferred brands are Atcom, Polycom and Grandstream. However any other good brand is also acceptable. Our client is interested in cheaper phones. Can anyone provide in Pakistan ?

Regards,

--
Kashif Naeem
Director
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: kashif@haditelecom.com
MSN: kashif__naeem@hotmail.com
Gmail: meet.kashif@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.

Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan

Senad,

Mind if I ask who that provider is?

Thanks.

Sent from my iPhone

On Dec 31, 2007, at 8:10 AM, Senad Jordanovic <senad@bicom.us> wrote:

> Justin Case wrote:
>> Tell me when to stop laughing. Multiple channels and unlimited
>> minutes ?
>> No sane person will give that to you.
>>
>
>
> Yap I agree...
>
> but but for about $900 per month one could get T1 (24 channels)
> unlimited in/out as far I seen last time our providers rates.
>
>
> Senad
>
>> On Dec 30, 2007 2:16 AM, Steve Finkelstein < sf@stevefink.net
>> <mailto:sf@stevefink.net>> wrote:
>>
>> Hi all,
>>
>> I have a budget to work with and was wondering if there are any
>> folks providing SIP/IAX2 trunking for unlimited inbound/outbound
>> for
>> a flat rate? We're in the budget range of roughly $5,000 a month
>> and
>> we need multiple channels per DID.
>>
>> I'm not sure if something like this is feasible in the world of
>> VoIP
>> -- and I only need to be able to make domestic/USA calls.
>>
>> Thanks for any potential leads.
>>
>> Happy holidays!
>>
>> - sf
>>
>> _______________________________________________
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>> digital.com--
>>
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>> To UNSUBSCRIBE or update options visit:
>>

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>>
>>
>>
>> ---
>> ---------------------------------------------------------------------
>>
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>
>
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Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan

Justin Case wrote:
> Tell me when to stop laughing. Multiple channels and unlimited minutes ?
> No sane person will give that to you.
>

Yap I agree...

but but for about $900 per month one could get T1 (24 channels)
unlimited in/out as far I seen last time our providers rates.

Senad

> On Dec 30, 2007 2:16 AM, Steve Finkelstein < sf@stevefink.net
> <mailto:sf@stevefink.net>> wrote:
>
> Hi all,
>
> I have a budget to work with and was wondering if there are any
> folks providing SIP/IAX2 trunking for unlimited inbound/outbound for
> a flat rate? We're in the budget range of roughly $5,000 a month and
> we need multiple channels per DID.
>
> I'm not sure if something like this is feasible in the world of VoIP
> -- and I only need to be able to make domestic/USA calls.
>
> Thanks for any potential leads.
>
> Happy holidays!
>
> - sf
>
> _______________________________________________
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>

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>
>
>
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Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan

Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you.

On Dec 30, 2007 2:16 AM, Steve Finkelstein < sf@stevefink.net> wrote:

Hi all,

I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID.

I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls.

Thanks for any potential leads.

Happy holidays!

- sf

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[asterisk-users] PRI Crapping Out Regularly

We have a server with a TE120 on a partial PRI trunk that several times
a day declares the PRI trunk down and stops handling calls until the
asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk
started.

Just before things go down, the log shows the following error:

ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500

at which point a "show pri spans" reports "PRI span 1/0: Provisioned,
Down, Active" and a "pri show span 1" reports:

Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 10000
T305 Timer: 30000
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3

(a) What is causing this?
(b) How can it be fixed?
(c) Why does Asterisk not recover automatically to what appears to be an
intermittent problem?

--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102

www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)

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Sunday, December 30, 2007

[asterisk-users] app_echo.c

hi, all
I have test echo application for just fun.
I can'nt understand why this is used below in .c file,

format = ast_best_codec(chan->nativeformats);
ast_set_write_format(chan, format);
ast_set_read_format(chan, format);

without this this application work fine.
then why this is used.

any suggestion??

Bhrugu mehta

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Re: [asterisk-users] asterisk callerid

Fixed.

While trying to locate the fault, I had added a fixed callerid to the
extension I am using to test. Once I fixed the original problem (dont
know what!), it still wouldnt work because of the fixed callerid

Feeling a bit ... ahhh ... stupid!

Thanks all,
BillK

On Sun, 2007-12-30 at 19:49 -0500, C F wrote:
> In fact the from user is making you the trouble.
>
> On Dec 30, 2007 7:48 PM, C F <shmaltz@gmail.com> wrote:
> > Take out the fromuser field, you don't need it.
> >
> >
> > On Dec 30, 2007 7:37 PM, William Kenworthy <billk@iinet.net.au> wrote:
> > > sip.conf:
> > >
> > > [iinet-PSTN]
> > > type=friend
> > > disallow=all
> > > allow=ulaw
> > > allow=g729
> > > allow=alaw
> > > host=dynamic
> > > port=5061
> > > username=iinet-PSTN
> > > fromuser=iinet-PSTN
> > > secret=yetta90
> > > context=incoming
> > > canreinvite=no
> > > nat=route
> > > qualify=yes
> > > insecure=very
> > >
> > > *Note - below I said that 9355nnnnn is the username - its actually the
> > > display name. The username is iinet-PSTN as above. I also just changed
> > > it to iinet-PSTN - still no callerid showing up at asterisk.
> > >
> > > BillK
> > >
> > > On Sun, 2007-12-30 at 10:06 -0500, C F wrote:
> > > > what does your sip.conf look like for the spa
> > > >
> > > > On 12/30/07, William Kenworthy <billk@iinet.net.au> wrote:
> > > > > I'm missing something simple I think:
> > > > >
> > > > > I have an spa3102 for which I want asterisk to use the incoming pstn
> > > > > callerid when it sends the call to a local extension (207).
> > > > >
> > > > > callerid works fine for the internal phones (between each other)
> > > > > The spa3102 is picking up the PSTN callerid and displays it in its
> > > > own
> > > > > status pages
> > > > >
> > > > > Asterisk however, doesnt see the callerid at all.
> > > > >
> > > > > The spa3102 is set to:
> > > > > PSTN CID For VoIP CID to Yes
> > > > > Dialplan 3 to (S0:<207>)
> > > > >
> > > > > In the SIP messages I can see the callerid as:
> > > > > From: MOBILE <sip:0419nnnnnn@192.168.1.1>;tag ...
> > > > > To: <sip:2-7@192.168.1.1>
> > > > >
> > > > > At the cli I get (The 935nnnnn is the user ID for the pstn)
> > > > >
> > > > > -- Executing NoOp("SIP/Main-08169b68", ""935nnnnn" <207>") in
> > > > new
> > > > > stack
> > > > > -- Executing Dial("SIP/Main-08169b68", "SIP/207|60|t") in new
> > > > stack
> > > > >
> > > > >
> > > > > Context is a basic 'catchall'
> > > > > [incoming]
> > > > > exten => s,1,NoOp(${CALLERID})
> > > > > exten => s,n,Dial(SIP/207,90,t)
> > > > > exten => s,n,Dial(SIP/202,90,t)
> > > > > exten => s,n,Congestion
> > > > > exten => s,n,Busy
> > > > > exten => s,n,Hangup
> > > > >
> > > > > What am I missing?
> > > > >
> > > > > BillK
> > > > >
> > > > >
> > > > > --
> > > > > William Kenworthy <billk@iinet.net.au>
> > > > > Home in Perth!
> > > > >
> > > > > _______________________________________________
> > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > > >
> > > > > asterisk-users mailing list
> > > > > To UNSUBSCRIBE or update options visit:
> > > > >

http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > >
> > > --
> > > William Kenworthy <billk@iinet.net.au>
> > > Home in Perth!
> > >
> >
--
William Kenworthy <billk@iinet.net.au>
Home in Perth!

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Re: [asterisk-users] asterisk callerid

sip.conf:

[iinet-PSTN]
type=friend
disallow=all
allow=ulaw
allow=g729
allow=alaw
host=dynamic
port=5061
username=iinet-PSTN
fromuser=iinet-PSTN
secret=yetta90
context=incoming
canreinvite=no
nat=route
qualify=yes
insecure=very

*Note - below I said that 9355nnnnn is the username - its actually the
display name. The username is iinet-PSTN as above. I also just changed
it to iinet-PSTN - still no callerid showing up at asterisk.

BillK

On Sun, 2007-12-30 at 10:06 -0500, C F wrote:
> what does your sip.conf look like for the spa
>
> On 12/30/07, William Kenworthy <billk@iinet.net.au> wrote:
> > I'm missing something simple I think:
> >
> > I have an spa3102 for which I want asterisk to use the incoming pstn
> > callerid when it sends the call to a local extension (207).
> >
> > callerid works fine for the internal phones (between each other)
> > The spa3102 is picking up the PSTN callerid and displays it in its
> own
> > status pages
> >
> > Asterisk however, doesnt see the callerid at all.
> >
> > The spa3102 is set to:
> > PSTN CID For VoIP CID to Yes
> > Dialplan 3 to (S0:<207>)
> >
> > In the SIP messages I can see the callerid as:
> > From: MOBILE <sip:0419nnnnnn@192.168.1.1>;tag ...
> > To: <sip:2-7@192.168.1.1>
> >
> > At the cli I get (The 935nnnnn is the user ID for the pstn)
> >
> > -- Executing NoOp("SIP/Main-08169b68", ""935nnnnn" <207>") in
> new
> > stack
> > -- Executing Dial("SIP/Main-08169b68", "SIP/207|60|t") in new
> stack
> >
> >
> > Context is a basic 'catchall'
> > [incoming]
> > exten => s,1,NoOp(${CALLERID})
> > exten => s,n,Dial(SIP/207,90,t)
> > exten => s,n,Dial(SIP/202,90,t)
> > exten => s,n,Congestion
> > exten => s,n,Busy
> > exten => s,n,Hangup
> >
> > What am I missing?
> >
> > BillK
> >
> >
> > --
> > William Kenworthy <billk@iinet.net.au>
> > Home in Perth!
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >

http://lists.digium.com/mailman/listinfo/asterisk-users
> >
--
William Kenworthy <billk@iinet.net.au>
Home in Perth!

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Re: [asterisk-users] Zap channels for HFC-S PCI card not responding

Quoting Tzafrir Cohen <tzafrir.cohen@xorcom.com>:

> What do you mean by "busy"? What exactly do you see?

This kind of thing:

# cat /proc/zaptel/*

Span 1: ZTHFC1 "HFC-S PCI A Zaptel Driver card 0 [TE]" (MASTER) AMI/CCS

1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Span 2: ZTHFC1 "HFC-S PCI A ISDN card 1 [TE]" AMI/CCS

4 ZTHFC1/0/1 Clear (In use)
5 ZTHFC1/0/2 Clear (In use)
6 ZTHFC1/0/3 HDLCFCS (In use)

Any attempts to call out result in the following CLI output:

[Dec 30 16:15:41] WARNING[12918]: app_dial.c:1130 dial_exec_full:
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1000-081ff9f8' status is 'CONGESTION'
[Dec 30 16:15:41] NOTICE[12918]: cdr.c:434 ast_cdr_free: CDR on
channel 'SIP/1000-081ff9f8' not posted

CLI> zap restart:
Destroying channels and reloading zaptel configuration.
== Parsing '/etc/asterisk/zapata.conf': Found
== Parsing '/etc/asterisk/zapata-channels.conf': Found
[Dec 30 16:32:41] WARNING[13612]: chan_zap.c:1081 zt_open: Unable to
specify channel 1: Device or resource busy
[Dec 30 16:32:41] ERROR[13612]: chan_zap.c:7501 mkintf: Unable to open
channel 1: Device or resource busy
here = 0, tmp->channel = 1, channel = 1
[Dec 30 16:32:41] ERROR[13612]: chan_zap.c:12266 build_channels: Unable
to register channel '1-2'
[Dec 30 16:32:41] WARNING[13612]: chan_zap.c:11554 zap_restart: Reload
channels from zap config failed!

This and more is from my previous message (sorry, that didn't just
contain configuration information).

Thanks,

Jaap

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Re: [asterisk-users] sip.conf for internetcalls.com

404 means that the number you are dialing is not available on the remote
end. Is there anything that you do that makes it break or is it random ? If
it is random I would speak to your ITSP.

----- Original Message -----
From: "Jaap Winius" <jwinius@umrk.to>
To: <asterisk-users@lists.digium.com>
Sent: Tuesday, December 25, 2007 1:14 AM
Subject: Re: [asterisk-users] sip.conf for internetcalls.com

> Quoting Justin Case <nogoodnameswereavailable@gmail.com>:
>
>> What comes up in the Asterisk CLI?
>
> When it's not working, nothing appears in the CLI even though I've used
> "set verbose 10".
>
>> Also it can be a NAT issue?
>
> How can that lead to this intermittent behavior? I've already set
> "nat=yes". Also, I'm using an ADSL router with a NAT; not anything
> like iptables.
>
>> Have Asterisk register every 3-4 minutes.
>
> I'm not sure how to do that. I found "defaultexpirey", but the default for
> it
> is two minutes. Anyway, why would that help with Asterisk, when my
> previous SIP client, a Linksys SPA3000, was configured with a register
> expire time of an
> hour and worked fine with InternetCalls.com.
>
> I think something else is going on. Using tcpdump, I see this when
> things are working okay:
>
> ----------------------
> 23:38:05.354523 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip >
> bitis.umrk.to.sip: SIP, length: 847
> INVITE sip:inetcalls-in@192.168.1.10:5060 SIP/2.0
> Via
> 23:38:05.355065 IP bitis.umrk.to.sip >
> 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip: SIP, length: 471
> FSIP/2.0 100 Trying
> Via: SIP/2.0/UDP 194.221.62.198:50
> .....
> 23:38:11.007350 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip >
> bitis.umrk.to.sip: SIP, length: 507
> ACK sip:inetcalls-in@192.168.1.10:5060 SIP/2.0
> Via: S
> ----------------------
>
> The ACK packet is sent after the conversation (.....) has ended.
> However, when it doesn't work, I see this:
>
> ----------------------
> 23:42:24.736377 IP 194.120.0.198.sip > bitis.umrk.to.sip: SIP, length: 841
> INVITE sip:inetcalls-in@192.168.1.10:5060 SIP/2.0
> Via
> 23:42:24.736898 IP bitis.umrk.to.sip > 194.120.0.198.sip: SIP, length: 445
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 194.120.0.198:
> 23:42:24.756967 IP 194.120.0.198.sip > bitis.umrk.to.sip: SIP, length: 505
> ACK sip:inetcalls-in@192.168.1.10:5060 SIP/2.0
> Via: S
> ----------------------
>
> In this case, the ACK follows immediately after the "404 Not Found".
>
> Cheers,
>
> Jaap
>
>
> _______________________________________________
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>

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Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

Thank you!

Will it come to 1.4.16.3 or 1.4.17?

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Sunday, December 30, 2007 5:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

On Sunday 30 December 2007 06:30:09 Mindaugas Kezys wrote:
> Just want to double check. When you are using this for IAX2 then first
> query is with 'dynamic', right?
>
> And after that when no peer is found other query(-ies) are executed which
> retrieves correct info about IAX2 user?
>
> I will have to test this myself. If it is correct - then problem could be
> only for SIP and less trouble to troubleshoot.
>
> Thanks for info.
>
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> MOR - Advanced Billing for Asterisk PBX
>
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anthony
> Messina Sent: Sunday, December 30, 2007 1:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
>
> On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote:
> > Currently we are using 1.4.15 which does not have such nasty BUG.
> >
> > When I will be free, I will try to review Asterisk sources to find a
> > problem and submit patch to this.
> >
> > From this case I see that not much people are using Asterisk Realtime
> > with newest Asterisk version.
> >
> > When holidays will end more and more people will start to complain
> > about this.
>
> i found that it did not affect my iax2 tunks (outbound peers) in mysql
> realtime, but it did affect the sip trunks (outbound peers) in realtime.

Please update to the latest SVN 1.4 -- this should have already been fixed.

--
Tilghman

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Re: [asterisk-users] Zap channels for HFC-S PCI card not responding

On Sun, Dec 30, 2007 at 04:48:39PM +0100, Jaap Winius wrote:
> Hi list,
>
> After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error
> messages related to my HFC-S PCI card disappeared, but now I can't
> access the card's resources because it always seems to be busy. Any
> idea why?

What do you mean by "busy"? What exactly do you see?

--
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[asterisk-users] Zap channels for HFC-S PCI card not responding

Hi list,

After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error
messages related to my HFC-S PCI card disappeared, but now I can't
access the card's resources because it always seems to be busy. Any
idea why?

Thanks,

Jaap

PS -- Below is some info regarding my configuration.

===========================

Zaptel version: 1.4.7 (incl. firmware and modules).
OS: Debian etch.

Loaded modules:

zaphfc 13660 1
vzaphfc 24984 1
zaptel 185956 9 xpp,zaphfc,vzaphfc
crc_ccitt 2560 1 zaptel

# cat /proc/zaptel/*

Span 1: ZTHFC1 "HFC-S PCI A Zaptel Driver card 0 [TE]" (MASTER) AMI/CCS

1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Span 2: ZTHFC1 "HFC-S PCI A ISDN card 1 [TE]" AMI/CCS

4 ZTHFC1/0/1 Clear (In use)
5 ZTHFC1/0/2 Clear (In use)
6 ZTHFC1/0/3 HDLCFCS (In use)

# ztcfg -vv

Zaptel Version: 1.4.7-Xorcom-trunk-r5178
Echo Canceller: MG2
Configuration
======================

SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: D-channel (Default) (Slaves: 06)

6 channels to configure.

/etc/asterisk/zapata-channels.conf after running "genzaptelconf -sd -c nl":

group=0,11
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 1-2
group=
context=default

group=0,12
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 4-5
group=
context=default

/etc/asterisk/zapata.conf (supposed to work in the Netherlands):

[trunkgroups]

[channels]
language=en
context=isdn-in
switchtype=euroisdn
pridialplan=dynamic
prilocaldialplan=local
nationalprefix = 0
internationalprefix = 00
overlapdial=yes
signalling=bri_cpe_ptmp
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=100
rxgain=4.5
txgain=-3
group=1
callgroup=1
pickupgroup=1
immediate=yes
#include zapata-channels.conf

Abbreviated /etc/asterisk/extensions.conf:

[globals]

[general]

[isdn-out]
exten => _X.,1,Dial(Zap/g0/${EXTEN}@channels,,r)

[internal]
exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()

[phones]
include => internal
include => isdn-out

Any attempts to call out result in the following CLI output:

CLI> zap show channels:

Chan Extension Context Language MOH Interpret
pseudo default en default
1 from-pstn en default
2 from-pstn en default
4 from-pstn en default
5 from-pstn en default

Not a good idea, since that results in...

CLI> zap show channels:

Chan Extension Context Language MOH Interpret

the channels disappearing altogether!

===========================

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Re: [asterisk-users] asterisk callerid

what does your sip.conf look like for the spa

On 12/30/07, William Kenworthy <billk@iinet.net.au> wrote:
> I'm missing something simple I think:
>
> I have an spa3102 for which I want asterisk to use the incoming pstn
> callerid when it sends the call to a local extension (207).
>
> callerid works fine for the internal phones (between each other)
> The spa3102 is picking up the PSTN callerid and displays it in its own
> status pages
>
> Asterisk however, doesnt see the callerid at all.
>
> The spa3102 is set to:
> PSTN CID For VoIP CID to Yes
> Dialplan 3 to (S0:<207>)
>
> In the SIP messages I can see the callerid as:
> From: MOBILE <sip:0419nnnnnn@192.168.1.1>;tag ...
> To: <sip:2-7@192.168.1.1>
>
> At the cli I get (The 935nnnnn is the user ID for the pstn)
>
> -- Executing NoOp("SIP/Main-08169b68", ""935nnnnn" <207>") in new
> stack
> -- Executing Dial("SIP/Main-08169b68", "SIP/207|60|t") in new stack
>
>
> Context is a basic 'catchall'
> [incoming]
> exten => s,1,NoOp(${CALLERID})
> exten => s,n,Dial(SIP/207,90,t)
> exten => s,n,Dial(SIP/202,90,t)
> exten => s,n,Congestion
> exten => s,n,Busy
> exten => s,n,Hangup
>
> What am I missing?
>
> BillK
>
>
> --
> William Kenworthy <billk@iinet.net.au>
> Home in Perth!
>
> _______________________________________________
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>
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> To UNSUBSCRIBE or update options visit:
>

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>

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Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

Please update to the latest SVN 1.4 -- this should have already been fixed.

--
Tilghman

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Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right?

And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user?

I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot.

Thanks for info.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anthony Messina
Sent: Sunday, December 30, 2007 1:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote:
> Currently we are using 1.4.15 which does not have such nasty BUG.
>
> When I will be free, I will try to review Asterisk sources to find a
> problem and submit patch to this.
>
> From this case I see that not much people are using Asterisk Realtime
> with newest Asterisk version.
>
> When holidays will end more and more people will start to complain
> about this.

i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime.

--
Anthony -

http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote:
> Currently we are using 1.4.15 which does not have such nasty BUG.
>
> When I will be free, I will try to review Asterisk sources to find a
> problem and submit patch to this.
>
> From this case I see that not much people are using Asterisk Realtime with
> newest Asterisk version.
>
> When holidays will end more and more people will start to complain about
> this.

i found that it did not affect my iax2 tunks (outbound peers) in mysql
realtime, but it did affect the sip trunks (outbound peers) in realtime.

--
Anthony -

http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

Currently we are using 1.4.15 which does not have such nasty BUG.

When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this.

From this case I see that not much people are using Asterisk Realtime with newest Asterisk version.

When holidays will end more and more people will start to complain about this.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anthony Messina
Sent: Sunday, December 30, 2007 12:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1

On Wednesday 19 December 2007 05:48:01 pm Mindaugas Kezys wrote:
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tilghman
> Lesher Sent: Thursday, December 20, 2007 1:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
>
> On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote:
> > [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql:
> > MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name =
> > 'Provider' AND host = 'dynamic'
> >
> > Note: host = 'dynamic'
>
> Correct, that's the FIRST lookup that is done.
>
> It then checks the IP address and does:
>
> "SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89'"
> where the IP address is what is sent in the SIP INVITE.
>
> If that fails, it does a lookup only on the name (old behavior).
>
> If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND
> port='5060'
>
> If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND
> port='5060'
>
> If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and
> checks every match for insecure=yes
>
> If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and
> checks every match for insecure=yes
>
> And if that fails, then it returns no match. So all of those queries
> had to run and fail for you to get no match.

were you ever able to get a solution for this? i seem the same problem when storing my sip trunks in mysql, using 1.4.16.2

--
Anthony -

http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [asterisk-users] Load Balancing over 2 E1 Lines

You can use the asterisk db for this. Simply set a variable to 1 or 0 if 1
set to 0 and use g2 if 0 set to 1 and use g1.
----- Original Message -----
From: "Andres Jimenez" <gandresin@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Wednesday, December 12, 2007 11:28 AM
Subject: Re: [asterisk-users] Load Balancing over 2 E1 Lines

> On Dec 12, 2007 8:08 AM, Eric Delaporte <edelaporte@gmx.de> wrote:
>
>
>> I read something about DIAL(Zap/r1/…) for using round robin, and it seems
>> to
>> work.
> That will give you the same number of calls routed to each line
>
>> Is there any other possible way to make sure that all lines are used in
>> the
>> same amount of minutes?
> You are going to need an AGI app or something storing how many minutes
> have been routed through each line and, on every call, choosing the
> less used one as the line to go out.
>
>
> --
> Andres Jimenez
>
> GPG : http://www.andresin.com/gpg/gandresin@gmail.com.asc
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>

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Saturday, December 29, 2007

[asterisk-users] asterisk callerid

I'm missing something simple I think:

I have an spa3102 for which I want asterisk to use the incoming pstn
callerid when it sends the call to a local extension (207).

callerid works fine for the internal phones (between each other)
The spa3102 is picking up the PSTN callerid and displays it in its own
status pages

Asterisk however, doesnt see the callerid at all.

The spa3102 is set to:
PSTN CID For VoIP CID to Yes
Dialplan 3 to (S0:<207>)

In the SIP messages I can see the callerid as:
From: MOBILE <sip:0419nnnnnn@192.168.1.1>;tag ...
To: <sip:2-7@192.168.1.1>

At the cli I get (The 935nnnnn is the user ID for the pstn)

-- Executing NoOp("SIP/Main-08169b68", ""935nnnnn" <207>") in new
stack
-- Executing Dial("SIP/Main-08169b68", "SIP/207|60|t") in new stack

Context is a basic 'catchall'
[incoming]
exten => s,1,NoOp(${CALLERID})
exten => s,n,Dial(SIP/207,90,t)
exten => s,n,Dial(SIP/202,90,t)
exten => s,n,Congestion
exten => s,n,Busy
exten => s,n,Hangup

What am I missing?

BillK

--
William Kenworthy <billk@iinet.net.au>
Home in Perth!

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Re: [asterisk-users] macports testing of asterisk

On Sat, Dec 29, 2007 at 05:31:45PM -0500, Marc Blanchet wrote:
> Hi,
> I recently submitted to the macports project a portfile enabling
> MacOSX users to use the simple macports system to install asterisk.

[snip]

Marc, thanks for your contributions.

One note: when you post a new message to this list please start a new
message, rather than replying to an existing message and editing the
subject line. Otherwise your message appears to be part of that thread.

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen@xorcom.com
+972-50-7952406 mailto:tzafrir.cohen@xorcom.com
http://www.xorcom.com

iax:guest@local.xorcom.com/tzafrir

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Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)

Dean,

Guess it is a moot point since Justin said it will be released under GPL
which makes his post totally inline with this list.

I am not sure what you do not understand about the list name, it is very
CLEARLY called "Asterisk Users Mailing List - *Non-Commercial
Discussion*", peddle your wares directly to the poster or on the biz
list, is it really that hard to understand?

Thanks,
Steve Totaro

Dean Collins wrote:
> Steve,
>
> Lol firstly I dont remember the exact wording I used but I'm fairly
> sure it was a request for people to post to the list with their
> features....(though I am accessing this email remotely so cant access
> my archives and it's not coming up in google) having said that I have
> often put my money where my mouth is and offered to invest in feature
> development before so cant be sure.
>
> Secondly - I've already posted the voicemail concept idea to the
> asterisk email list that i had for a better voicemail to email
> interface before about a year ago basically it's delivering a
> 'player' in a html format so you can play the voicemail in
> outlook/gmail etc - oh and way before Apple iphone came up with the
> idea of visual voicemail.
>
> (edit: I went back and found a link - Nov 9th 2006....guess no one
> thought it was a good idea back then or had the desire/funds to pay
> for it to be built
> http://lists.digium.com/pipermail/asterisk-users/2006-November/171995.html guess
> thats the problem with open source development.....maybe you want to
> build this for freee Steve....?)
>
> All I was suggesting was that if Justin's new application was being
> offered on a commercial basis (it's not - it's GPL) that i would be
> happy to spend time scoping out the application in return for
> consulting fees/equity the same way I have helped a number of
> commercial and non commercial application developers in the Asterisk
> community.
>
> As open source is all about choice....feel free to choose not to reply
> to any other idea discussions I may post on the list..... us 'sharks'
> dont care.
>
> If anyone bothered downloading the mp3 of fridays conference call you
> would hear pretty much most of what I want to say about the balance
> between commercial and non-commercial input into the asterisk
> community and it's future development.
>
>
>
> Cheers,
> Dean
>
>
> ------------------------------------------------------------------------
> *From:* asterisk-users-bounces@lists.digium.com on behalf of Steve Totaro
> *Sent:* Fri 12/28/2007 1:07 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] New voicemail app (supports many
> interfaces, including Audix)
>
> Dean,
>
> I am saying nothing of the sort.
>
> To clarify, I am saying that I do not see the people you mentioned
> fishing for free ideas or posting commercially to the User's list with
> the exception of yourself now, when you had affiliation with Mexuar, and
> a handful of other times.
>
> I find it funny when I see the commercials on TV and email spam asking
> for "Inventor's Ideas", all it would take is one sucker to rip off with
> a great idea to make all those commercials pay off.
>
> Some of us have long memories, can put pieces together and will call you
> out when something is fishy.
>
> Thanks,
> Steve Totaro
>
> Dean Collins wrote:
> > So you're saying people like snapanumber, mexuar and other commercially
> > related Asterisk applications cant charge money huh Steve?
> >
> > Maybe this conference call may interest you.
> > http://recordings.talkshoe.com/TC-22622/TS-75263.mp3
> >
> >
> > Cheers,
> > Dean
> >
> >
> >
> >> -----Original Message-----
> >> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> >> bounces@lists.digium.com] On Behalf Of Steve Totaro
> >> Sent: Thursday, 27 December 2007 7:40 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] New voicemail app (supports many
> >>
> > interfaces,
> >
> >> including Audix)
> >>
> >> Licensing your thoughts, do you have a unique patent or a even a
> >>
> > patent
> >
> >> on an improvement?
> >>
> >> Aren't you the guy soliciting the user's list for "The Next Geewhiz
> >>
> > App"
> >
> >> idea a while ago? Sharks are everywhere.
> >>
> >> Anyways, this is the Users, soliciting should be done on the Biz list.
> >>
> >> Thanks,
> >> Steve Totaro
> >>
> >> Dean Collins wrote:
> >>
> >>> Are you selling/licensing the new voicemail app or just asking if
> >>> people want to download it?
> >>>
> >>>
> >>>
> >>> The reason for asking is if you are selling it I have some thoughts
> >>>
> > on
> >
> >>> how voicemail on asterisk can be improved and would like to discuss
> >>> licensing this to you.
> >>>
> >>>
> >>>
> >>> Not really working for the next few days till after new year though
> >>>
> > so
> >
> >>> email replies will be sporadic.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> Cheers,
> >>>
> >>> Dean
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> > ------------------------------------------------------------------------
> >
> >>> *From:* asterisk-users-bounces@lists.digium.com
> >>> [mailto:asterisk-users-bounces@lists.digium.com] *On Behalf Of
> >>>
> > *Justin
> >
> >>> Newman
> >>> *Sent:* Thursday, 27 December 2007 5:38 PM
> >>> *To:* asterisk-users@lists.digium.com
> >>> *Cc:* nt_jnewman@yahoo.com
> >>> *Subject:* [asterisk-users] New voicemail app (supports many
> >>> interfaces,including Audix)
> >>>
> >>>
> >>>
> >>> We just completed a new implementation of voicemail for Asterisk.
> >>>
> > It's
> >
> >>> much cleaner than Comedian mail and can emulate several voicemail
> >>>
> > user
> >
> >>> interfaces, including Audix. It's a great replacement for Audix. All
> >>> of the sounds/prompts are presently being re-recorded by a
> >>> professional female voice.
> >>>
> >>> If you are interest in the app, let us know at nt_jnewman@yahoo.com.
> >>>
> >>> Justin
> >>>
> >>>
> >>>
> >>>
> >>>
> > ------------------------------------------------------------------------
> >
> >>> Looking for last minute shopping deals? Find them fast with Yahoo!
> >>> Search.
> >>>
> >>>
> > <http://us.rd.yahoo.com/evt=51734/*http:/tools.search.yahoo.com/newsearc
> > h/
> >
> >> category.php?category=shopping>
> >>
> >>>
> > ------------------------------------------------------------------------
> >
> >>
> >>
> >
>
>
> _______________________________________________
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