[asterisk-users] asterisk 1.2 or 1.4 for conference call service

Hi:
I want to have conference call service and I have A102d sangoma's card so I install asterisk 1.2.x or 1.4.x?
Best regards.

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[asterisk-users] A102d sangoma's card and ztdummy

Hi:
I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference?
Best regards.

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[asterisk-users] OT - How to script softphone installation

Hello,

Before diving into this, I would like to gather your opinion about the capability of software such as http://nsis.sourceforge.net/Main_Page to used to automate softphone installation on Windows platforms.

Is it possible to use this to push softphone installations and uninstallations ?
Is it a long process to hide a specific softphone installation process into an "NSIS controlled" one ?

Ideas are welcome.

regards

Re: [asterisk-users] Problems with Polycom 300/500/600

BJ Weschke wrote:
> On 8/31/07, Joe Acquisto <joea@j4computers.com> wrote:
>> Any great disadvantage to using polycom 300/500/600 vs the 301/501/601?
>>
>
> I recall reading in the release notes of the latest release of the
> firmware (2.2+) that I believe they've finally stopped supporting the
> earlier models so it looks like you are reaching or have reached an
> EOL period on firmware with those models. Aside from that, if you're
> happy with current functionality of those phones as they stand now,
> they'll probably be fine for quite some time to come.
>

I believe they only dropped the 300 and possibly the 500 with the latest
firmware (per the release notes for 2.2.0). The 600 already had the
increased memory on the phone (which was the major difference from the
300/301 and 500/501.

That being said, the 300, 500 and 600 are discontinued. You might be
able to get them cheap, but I wouldn't use them in a new install.

Darrick
--
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[asterisk-users] Problems with Asterisk 1.2.23 and Polycom 601

Hi,

 

I have a polycom 601 with 3 expansion modules attached and about 40 extensions.  When someone does a “page all” from that phone, sometimes the expansion modules reboot and sometimes the phone itself reboots.  This happens randomly.  I suspected it to be a problem with the presence / buddy watch feature in the directory on the phone so I turned that off for all the extensions listed in the directory and the problem has not re-occurred.  However, this means the status of the other phones cannot be seen on the expansion modules of the reception phone and I am interested in that.

 

Any pointers?  I have done a couple of searches and not come up with anything concrete on resolving this problem.  I am running Asterisk 1.2.23. Any help would be appreciated.

 

Regards,

 

Joseph

Re: [asterisk-users] How to handle "+" prefix

So yeah, I can admit that this is all true outside the US. ^^, sorry.

Steve Kennedy wrote:
> On Fri, Aug 31, 2007 at 10:03:07AM -0600, Kai-Uwe Jensen wrote:
>
>
>> On 8/31/07, Anthony Francis <anthonyf@rockynet.com> wrote:
>>
>>> Mindfully wanting to use a + instead of knowing the international access code seems like willful ignorance to me.
>>>
>> I beg to differ. Consider cell phones as an example. They all provide
>> + keys. And it is considered a best practice to store phone numbers in
>> address books as "+ country_code area_code number" so that you can
>> change locations (ever traveled to Europe?) easily, without having to
>> reprogram all your contacts' numbers.
>>
>
> Also all SMS are sent in GSM international format (i.e. all prefixed
> with +CC).
>
>
> Steve
>
>

--
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
voip@rockynet.com

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Re: [asterisk-users] How to handle "+" prefix

I knew that was true about GSM networks outside of the US, but to be
honest, I am not concerned with those networks ^^.

Mr Shunz wrote:
> On 8/31/07, Anthony Francis <anthonyf@rockynet.com> wrote:
>
>> I don't know if you intended to be rude with the "normal
>> human" comment but it sure seems like it when reading your
>> reply. Also how many users know they can dial ** to get a +?
>> Especially when so many cannot as said earlier be relied upon
>> to know their country access code. I guess my point is if
>> there isn't a + key on the phone, and on most there is not,
>> then it simply shouldn't be used. Mindfully wanting to use a +
>> instead of knowing the international access code seems like
>> willful ignorance to me.
>>
>
> Actually here in Italy cell phone companies send you the caller
> id ALWAYS with the international prefix (as +39) so i (and every
> one i know) have the address book filled with numbers like
> +393....... Having tested various GSM + WiFi + SIP phones
> we had to take care of this case
>
> actually we use to call only Italy so we have smth like:
>
> exten => _+39.,1,Macro(dialCell)
>
> and
>
> [macro-dialCell]
> exten => s,1,Dial(OH323/${MACRO_EXTEN:3}@${GATEWAY},,TW)
>
> cheers
>
>

--
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
voip@rockynet.com

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Re: [asterisk-users] Cisco 7960 Won'

Jason wrote:
> Dan Austin wrote:
>> Shawn wrote:
>>> I'm having a wierd problem with a Cisco 7960 (sccp2)
>>> and asterisk (1.4.2)
>
>>> If the call that I'm trying to make goes through,
>>> everything works fine. But if there's any sort of
>>> error (like me messing around in my extensions.conf,
>>> etc). I can't get the connection to drop. ie: If I get
>>> the conjestion tone and hang up the phone, I can do a
>>> sccp show channels I can see that the channel is still
>>> in use (even after several minutes). If I pick up the
>>> phone to attempt to make another call, I get an error
>>> that it can't put the current call on hold to start
>>> the new call.
>
>>> What am I missing?
>> An upgrade.
>>
>> The sccp channel in early 1.4 had quite a number of problems,
>> and it was completely broken in 1.4.3 to 1.4.6
>>
>> Any version after 1.4.7 should work better, with the latest
>> being the best choice.
>>
>> Dan
>

> Well, he's also using chan_sccp, so no amount of upgrading
> is going to help with that.

> In my opinion (and I think Dan and several others would agree),
> chan_skinny is far more stable (and active...) than chan_sccp.

Bugger! I should have noted the 'sccp show channels' command.
I tend to swap skinny/SCCP automatically, since Cisco uses
both in the documentation, and had it in my head that he
meant skinny....

Yes, chan_skinny in 1.4.7+ has had major love applied. I only
have a couple test phones hooked up for development, so my
impression of stability is not worth much, but I think we
have managed to fix up the most hideous bugs.

If we can keep up the pace, chan_skinny in 1.6 is going to rock.

Sorry for the confusion.

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Re: [asterisk-users] Problems with Polycom 300/500/600

On 8/31/07, Joe Acquisto <joea@j4computers.com> wrote:
> Any great disadvantage to using polycom 300/500/600 vs the 301/501/601?
>

I recall reading in the release notes of the latest release of the
firmware (2.2+) that I believe they've finally stopped supporting the
earlier models so it looks like you are reaching or have reached an
EOL period on firmware with those models. Aside from that, if you're
happy with current functionality of those phones as they stand now,
they'll probably be fine for quite some time to come.

--
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[asterisk-users] Problems with Polycom 300/500/600

Any great disadvantage to using polycom 300/500/600 vs the 301/501/601?

joe a.

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[asterisk-users] Cisco 7960 sccp

What is involved in getting SIP firmware into a Cisco 7960 with sccp installed?

Expensive image from Cisco? Plated in unobtanium?

joe a.

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Re: [asterisk-users] VoIP+IM with Asterisk+Jabber

On 8/31/07, demuel@thephinix.org <demuel@thephinix.org> wrote:
> Ows, I suppose that * can only do c2s to google talk to which I did and I got audio both
> ways. Yet I have not seen anything so far how * could do a s2s to google talk.

Indeed, the Jabber module was not designed to make Asterisk a Jabber server.

Philippe

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Re: [asterisk-users] Cisco 7960 Won'

On 13:52, Fri 31 Aug 07, Jason Parker wrote:
> Dan Austin wrote:
> > Shawn wrote:
> >> I'm having a wierd problem with a Cisco 7960 (sccp2)
> >> and asterisk (1.4.2)
> >
> >> If the call that I'm trying to make goes through,
> >> everything works fine. But if there's any sort of
> >> error (like me messing around in my extensions.conf,
> >> etc). I can't get the connection to drop. ie: If I get
> >> the conjestion tone and hang up the phone, I can do a
> >> sccp show channels I can see that the channel is still
> >> in use (even after several minutes). If I pick up the
> >> phone to attempt to make another call, I get an error
> >> that it can't put the current call on hold to start
> >> the new call.
> >
> >> What am I missing?
> > An upgrade.
> >
> > The sccp channel in early 1.4 had quite a number of problems,
> > and it was completely broken in 1.4.3 to 1.4.6
> >
> > Any version after 1.4.7 should work better, with the latest
> > being the best choice.
> >
> > Dan
> >
>
> Well, he's also using chan_sccp, so no amount of upgrading is going to help
> with that.
>
> In my opinion (and I think Dan and several others would agree), chan_skinny is
> far more stable (and active...) than chan_sccp.

as on of the 'several outhers' I totally agree.
We used to run chan_sccp for our kirk setup and some cisco
phones.
The switch to chan_skinny made everything usable again :)
The random crashes and lockups you get with chan_sccp are
too annoying :)
--

Michiel van Baak
michiel@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [asterisk-users] OT: DELL Platforms

Dovid B wrote:
> ----- Original Message -----
> From: "Steve Totaro" <stotaro@first-notification.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Sent: Tuesday, August 28, 2007 11:51 AM
> Subject: Re: [asterisk-users] OT: DELL Platforms
>
>
>
>> Dovid B wrote:
>>
>>> <snip>
>>>
>>>
>>>> I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM
>>>> and a couple 250gig SATA drives. Totally VoIP so I cannot comment on
>>>> cards or interrupts, but so far it has been flawless.
>>>>
>>>> I would like to see how many G729/ULAW conversions it could handle. How
>>>> would I go about benchmarking that?
>>>>
>>>> Thanks,
>>>> Steve
>>>>
>>>>
>>> </snip>
>>>
>>> I would advise against the SC Series because there is no RAC card option.
>>> You never know that you needed one till you need to format a box from
>>> your
>>> hotel room ;)
>>>
>>> I have used some Poweredge 1850's with Asterisk (VOIP Only) and I have
>>> been
>>> real happy.
>>>
>>>
>>>
>> Not so much a problem since the CoLo is staffed 24/7 and glad to help.
>> Soon to add KVMoIP.
>>
>> There is a rudementary (have not played with it yet)
>> /The latest industry-standard Intelligent Platform Management Interface
>> (IPMI) 2.0 Baseboard Management Controller (BMC) allows remote,
>> out-of-band management over a network or serial connection with any
>> industry-standard IPMI management program.
>>
>> /I will play with this feature a bit before investing in KVMoIP.
>>
>> Thanks,
>> Steve Totaro
>>
>
> Steve,
> You have a point but you can't cycle the power with KVM over VOIP. The DC is
> staffed 24/7 but it takes time till they pick up, reboot the server etc.
> Also who wants to pay them to format a box if I can do it remotely ;)
>
>
That is why I have web switches. Pricey for what they do but well worth
it when you need it, plus they are very well constructed.

http://www.controlbyweb.com/webswitch/index.html

Thanks,
Steve Totaro

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Re: [asterisk-users] VoIP+IM with Asterisk+Jabber

Ows, I suppose that * can only do c2s to google talk to which I did and I got audio both
ways. Yet I have not seen anything so far how * could do a s2s to google talk.

> Hi Alejandro,
>
> the Jabber module in Asterisk is available starting from the 1.4
> series. Therefore, you can connect Asterisk as a client (or component)
> to your Jabber server after you've upgraded to 1.4.
>
> You'll get detailed information here :
> http://www.voip-info.org/wiki-Asterisk+Jabber
> http://www.voip-info.org/wiki/view/Asterisk+Google+Talk
>
> Philippe
>
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>

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>

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Re: [asterisk-users] Latency, Jitter and Lost packets...

On Fri, 2007-08-31 at 15:51 -0300, Luis Antonio Prata Barbosa wrote:
> Does anybody know any software that give me Latencty, Jitter and Lost
> packets to analyze my Call quality ???

The packet sniffer called Wireshark has a great RTP analysis tool that
will show you (and even graph!) the latency, jitter, lost packets, and
out-of-order packets. It slices! It dices! And it does a great job at
making an Asterholic's life much easier. :-) Oh, and did I mention it's
open source!

--
Jared Smith
Community Relations Manager
Digium, Inc.

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Re: [asterisk-users] Latency, Jitter and Lost packets...

Ping... =)

>From: "Luis Antonio Prata Barbosa" <luispratalistas@gmail.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial
>Discussion<asterisk-users@lists.digium.com>
>To: "Asterisk Users Mailing List - Non-Commercial
>Discussion"<asterisk-users@lists.digium.com>
>Subject: [asterisk-users] Latency, Jitter and Lost packets...
>Date: Fri, 31 Aug 2007 15:51:09 -0300
>
>Hi,
>
>Does anybody know any software that give me Latencty, Jitter and Lost
>packets to analyze my Call quality ???
>
>
>Luis

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_________________________________________________________________
Descubra como mandar Torpedos do Messenger para o celular!

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Re: [asterisk-users] Cisco 7960 Won'

Dan Austin wrote:
> Shawn wrote:
>> I'm having a wierd problem with a Cisco 7960 (sccp2)
>> and asterisk (1.4.2)
>
>> If the call that I'm trying to make goes through,
>> everything works fine. But if there's any sort of
>> error (like me messing around in my extensions.conf,
>> etc). I can't get the connection to drop. ie: If I get
>> the conjestion tone and hang up the phone, I can do a
>> sccp show channels I can see that the channel is still
>> in use (even after several minutes). If I pick up the
>> phone to attempt to make another call, I get an error
>> that it can't put the current call on hold to start
>> the new call.
>
>> What am I missing?
> An upgrade.
>
> The sccp channel in early 1.4 had quite a number of problems,
> and it was completely broken in 1.4.3 to 1.4.6
>
> Any version after 1.4.7 should work better, with the latest
> being the best choice.
>
> Dan
>

Well, he's also using chan_sccp, so no amount of upgrading is going to help
with that.

In my opinion (and I think Dan and several others would agree), chan_skinny is
far more stable (and active...) than chan_sccp.

--
Jason Parker
Digium

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[asterisk-users] Latency, Jitter and Lost packets...

Hi,

 

Does anybody know any software that give me Latencty, Jitter and Lost packets to analyze my Call quality ???

 

 

Luis

[asterisk-users] Cisco Directory Format

Hi Gang,

A little off topic (sorry..:) ) but anyone know what format Cisco phones
use for their contact dirctories. I want to set up my contact lists on
the phone, and cannot seem to get any info on it. I am working with a
7970 on Asterisk 1.4.8.

Thanks a bunch.
Kristian

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Re: [asterisk-users] E1 to Ethernet Bridge

Jared Smith wrote:
> On Fri, 2007-08-31 at 12:55 +0000, Arinze Izukanne wrote:
>
>> I am trying to Bridge 2 E1 interfaces over a long distance link
>>
>
> It's really not too difficult... it ends up looking like this:
>
> E1 -> Asterisk A -> IAX2 -> Asterisk B -> E1
>
> Asterisk box A is configured to take calls from the E1, and send the
> calls over IAX2 to box B. Box B takes calls in over IAX2, and sends
> them out its E1. (The reverse can also be setup as well, so that calls
> coming into box B over its E1 can be forwarded over IAX2 to box A, and
> so forth.)
>
> Let me know if you need me to go into further detail.
>
>
Assuming the link you are speaking of is an IP link, then the above
would work.

I find that SIP is much more reliable, try IAX with trunking and if you
have problems, give SIP a try and see if that solves your issue.

Thanks,
Steve

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[asterisk-users] Which IP Phone is really the best?

I need to quote a client for a job and I was just wondering.

Out of all the IP Phones out there, which one is the best and why?

Thank you all, all opinions will be accepted.

 

 

William Herrera

LAN/WAN Technical Consultant

Re: [asterisk-users] E1 to Ethernet Bridge

On Fri, Aug 31, 2007 at 12:55:24PM +0000, Arinze Izukanne wrote:
> Hello,
>
> I am trying to Bridge 2 E1 interfaces over a long distance link
> exactly the same way Redfone does. How can asterisk be configured
> to do that?

If I understand correctly, yyou should configure those two E1 interfaces
in Zaptel alone (and don't run Asterisk) and then expose them as
dynamic spans (ztd-eth ).

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

http://www.xorcom.com

iax:guest@local.xorcom.com/tzafrir

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[asterisk-users] AEL missing in recent 1.2 releases?

Greetings list,

I've just been upgrading one of our servers from 1.2.17 to 1.2.21.1-r1, and noticed that it's not picking up any of my macros written in AEL.

Upon further examination, it looks like pbx_ael is missing. Is this a deliberate change, or is this something I need to address in the pre-compile configuration?

Regards,

Chris
--
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Re: [asterisk-users] How to handle "+" prefix

On Fri, Aug 31, 2007 at 10:03:07AM -0600, Kai-Uwe Jensen wrote:

> On 8/31/07, Anthony Francis <anthonyf@rockynet.com> wrote:
> > Mindfully wanting to use a + instead of knowing the international access code seems like willful ignorance to me.
> I beg to differ. Consider cell phones as an example. They all provide
> + keys. And it is considered a best practice to store phone numbers in
> address books as "+ country_code area_code number" so that you can
> change locations (ever traveled to Europe?) easily, without having to
> reprogram all your contacts' numbers.

Also all SMS are sent in GSM international format (i.e. all prefixed
with +CC).

Steve

--
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UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net
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Re: [asterisk-users] Cisco 7960 Won'

Shawn wrote:
> I'm having a wierd problem with a Cisco 7960 (sccp2)
> and asterisk (1.4.2)

> If the call that I'm trying to make goes through,
> everything works fine. But if there's any sort of
> error (like me messing around in my extensions.conf,
> etc). I can't get the connection to drop. ie: If I get
> the conjestion tone and hang up the phone, I can do a
> sccp show channels I can see that the channel is still
> in use (even after several minutes). If I pick up the
> phone to attempt to make another call, I get an error
> that it can't put the current call on hold to start
> the new call.

> What am I missing?
An upgrade.

The sccp channel in early 1.4 had quite a number of problems,
and it was completely broken in 1.4.3 to 1.4.6

Any version after 1.4.7 should work better, with the latest
being the best choice.

Dan

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[asterisk-users] chan_sip.c:5495 sip_reg_timeout: ERROR

Hello,

 

I’ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages:

 

Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout:    -- Registration for 'user@sipserver' timed out, trying again (Attempt #19)

 

All trunks and extensions went to:

 

sipserver:5060      user         120 Request Sent

 

011                        (Unspecified)    D   N      0        UNKNOWN

 

Using ngrep I can see incoming messages to the server (port 5060), but no reponses from the server:

 

172.16.4.190:5060 -> sipserver:5060

  REGISTER sip: sipserver

 

But if I use CLI I do not see these incoming messages.

 

Any help on this?

 

Regards,

 

Reggie Arévalo

 

 

Re: [asterisk-users] Strange behaviour on Asterisk 1.4.9 with Queues...

On Fri, 2007-08-31 at 11:38 -0500, Carlos Chavez wrote:
> I am having a strange problem with an Asterisk server that has a small
> 5 seat call center. While everything seems to be working properly I if
> do a "core show channels" the server goes into a loop:

I'm not sure what might cause this, but I suggest you try it with
Asterisk 1.4.11 and see if the problem continues. There have been a
fairly large number of bugs fixed since 1.4.9 was released, and this
problem may have already been solved.

If you continue to have the same problem with 1.4.11, please open a bug
report on the bug tracker (http://bugs.digium.com/) and an Asterisk
developer will take a look at the problem as quickly as they can.

--
Jared Smith
Community Relations Manager
Digium, Inc.

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Re: [asterisk-users] How to handle "+" prefix

Anthony Francis wrote:
> SIP wrote:
>
> (many of our users do that, and they just type a + like a normal human)
>
> I don't know if you intended to be rude with the "normal human" comment but it sure seems like it when reading your reply. Also how many users know they can dial ** to get a +? Especially when so many cannot as said earlier be relied upon to know their country access code. I guess my point is if there isn't a + key on the phone, and on most there is not, then it simply shouldn't be used. Mindfully wanting to use a + instead of knowing the international access code seems like willful ignorance to me.
>
>
>
> _______________________________________________
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>
Woah. Simmer down there, Tex. That wasn't intended to be rude. I'm
merely stating that it doesn't require the use of complex menuing
systems when dialing from a web interface. Don't make an EFFORT to be
offended.

N.

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[asterisk-users] Strange behaviour on Asterisk 1.4.9 with Queues...

I am having a strange problem with an Asterisk server that has a small
5 seat call center. While everything seems to be working properly I if
do a "core show channels" the server goes into a loop:

pbxinsol*CLI> core show channels
Channel Location State
Application(Data)
Agent/4 133@default:1 Up Bridged
Call(UniCall/12-1)
SIP/133-094ba940 133@default:1 Up
(None)
UniCall/12-1 s@menu-henkel:5 Up
Queue(henkel)
Agent/2 110@default:1 Up Bridged
Call(UniCall/8-1)
SIP/110-094a7888 110@default:1 Up
(None)
UniCall/8-1 s@menu-henkel:5 Up
Queue(henkel)
Agent/1 136@default:1 Up Bridged
Call(UniCall/11-1)
SIP/136-09472298 136@default:1 Up
(None)
UniCall/11-1 s@menu-henkel:5 Up
Queue(henkel)
Agent/4 133@default:1 Up Bridged
Call(UniCall/12-1)
SIP/133-094ba940 133@default:1 Up
(None)
UniCall/12-1 s@menu-henkel:5 Up
Queue(henkel)
Agent/2 110@default:1 Up Bridged
Call(UniCall/8-1)
SIP/110-094a7888 110@default:1 Up
(None)
UniCall/8-1 s@menu-henkel:5 Up
Queue(henkel)
Agent/1 136@default:1 Up Bridged
Call(UniCall/11-1)
SIP/136-09472298 136@default:1 Up
(None)
UniCall/11-1 s@menu-henkel:5 Up
Queue(henkel)
...

I have to type exit to get out of Asterisk because there is no way to
stop, it keeps looping through the channels. This does not seem to
affect the operation of Asterisk though. Any ideas?

--
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001

Re: [asterisk-users] How long to detect an "h" exten?

What type of delays, when - after what command in the dialplan - do
you see the delay What exaclty are you trying to do in the dialplan?

On 8/31/07, Gavin Henry <gavin.henry@gmail.com> wrote:
> On 30/08/2007, C F <shmaltz@gmail.com> wrote:
> > Can you explain this question?
> > Just to clearify, exten => h will execute as soon as Asterisk is aware
> > that the channel was hung up. While app_hangup will execute a hangup
> > on an active channel.
>
> I'm just trying to track down some delays in my dialplans and wondered
> if this might the problem in hanging up a zap call.
>
> >
> >
> > On 8/30/07, Gavin Henry <gavin.henry@gmail.com> wrote:
> > > Dear All,
> > >
> > > How long should it take before a exten => h,1,Hangup() kicks in,
> > > versus a exten => s,n,Hangup()
> > >
> > > I'm just about to test, but thought I'd ask.
> > >
> > > --
> > > http://www.suretecsystems.com/services/openldap/
> > >
> > > _______________________________________________
> > > --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
> > >
> >
> > _______________________________________________
> > --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
> >
>
>
> --
> http://www.suretecsystems.com/services/openldap/
>
> _______________________________________________
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>

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Re: [asterisk-users] VoIP+IM with Asterisk+Jabber

Hi Alejandro,

the Jabber module in Asterisk is available starting from the 1.4
series. Therefore, you can connect Asterisk as a client (or component)
to your Jabber server after you've upgraded to 1.4.

You'll get detailed information here :
http://www.voip-info.org/wiki-Asterisk+Jabber
http://www.voip-info.org/wiki/view/Asterisk+Google+Talk

Philippe

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Re: [asterisk-users] How to handle "+" prefix

i live in the US, but travel.  i will second this ... Vodafone in Europe will give you a discount when you dial with a '+' instead of the local access code.  Probably because Europeans roam in many foreign countries each with a different access code for long distance dialing.  That would be messy, detect which country the user is calling in before pattern matching the phone # ... takes time and slows down calls per minute.  '+' dialing should be supported out of the box.

Putting a '+' in your Yahoo address book and then syncing that to your phone is easy.  That way, i can always dial that +0353 from whichever country i happen to be in at the time.

On 8/31/07, Mr Shunz <mrshunz@gmail.com> wrote:

On 8/31/07, Anthony Francis <anthonyf@rockynet.com> wrote:
>
> I don't know if you intended to be rude with the "normal
> human" comment but it sure seems like it when reading your
> reply. Also how many users know they can dial ** to get a +?
> Especially when so many cannot as said earlier be relied upon
> to know their country access code. I guess my point is if
> there isn't a + key on the phone, and on most there is not,
> then it simply shouldn't be used. Mindfully wanting to use a +
> instead of knowing the international access code seems like
> willful ignorance to me.

Actually here in Italy cell phone companies send you the caller
id ALWAYS with the international prefix (as +39) so i (and every
one i know) have the address book filled with numbers like
+393....... Having tested various GSM + WiFi + SIP phones
we had to take care of this case

actually we use to call only Italy so we have smth like:

exten => _+39.,1,Macro(dialCell)

and

[macro-dialCell]
exten => s,1,Dial(OH323/${MACRO_EXTEN:3}@${GATEWAY},,TW)

cheers

--
Daniele Santi        .o.
MrShunz@gmail.com    ..o
Linux User #415108   ooo

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Re: [asterisk-users] How to handle "+" prefix

On 8/31/07, Anthony Francis <anthonyf@rockynet.com> wrote:

> Mindfully wanting to use a + instead of knowing the international access code seems like willful ignorance to me.

I beg to differ. Consider cell phones as an example. They all provide
+ keys. And it is considered a best practice to store phone numbers in
address books as "+ country_code area_code number" so that you can
change locations (ever traveled to Europe?) easily, without having to
reprogram all your contacts' numbers.

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Re: [asterisk-users] How long to detect an "h" exten?

On 30/08/2007, C F <shmaltz@gmail.com> wrote:
> Can you explain this question?
> Just to clearify, exten => h will execute as soon as Asterisk is aware
> that the channel was hung up. While app_hangup will execute a hangup
> on an active channel.

I'm just trying to track down some delays in my dialplans and wondered
if this might the problem in hanging up a zap call.

>
>
> On 8/30/07, Gavin Henry <gavin.henry@gmail.com> wrote:
> > Dear All,
> >
> > How long should it take before a exten => h,1,Hangup() kicks in,
> > versus a exten => s,n,Hangup()
> >
> > I'm just about to test, but thought I'd ask.
> >
> > --
> > http://www.suretecsystems.com/services/openldap/
> >
> > _______________________________________________
> > --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
> >
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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>

--

http://www.suretecsystems.com/services/openldap/

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[asterisk-users] Sipp scenario for asterisk sip

Hey
I'm looking for an advanced scenario for sipp, that can be used for testing asterisk. Mainly I'm interested in making random calls between sipp pseudo-users. Did anyone try to do something like this?
Or has anyone got an example scenario with working loops?

Thanks

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Re: [asterisk-users] TC400B and show transcoder

Ben Dinnerville wrote:

> The document that describes the "show transcoder" command is the pdf on
> the digium website -
>
> http://www.digium.com/elqNow/elqRedir.htm?ref=http://www.digium.com/docs/TC400B/TC400B-user-manual.pdf

I'll get that corrected.

> As mentioned, the command does exist (only available when a card is
> present) on 1.2 but not 1.4 - if it is not meant to be in 1.4 i would
> recommend getting it in there, it is nice to be able to know what is
> happening on the system

I've just added this command to Asterisk 1.4; I don't know how it got
missed. Thanks for bringing it to our attention.

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

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[asterisk-users] app_conference

Is app_conference designed only for 1.4? I tried compiling against 1.2.24
and but get a no such file while looking for autoconf.h which is a file only
used in 1.4... anybody running app_conference on 1.2?

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Re: [asterisk-users] Channel banks for E1

Why don´t you try astribank32 ??? 

2007/8/30, Jan Marek <jmarek@jcu.cz>:

Hello all,

please, can anyone advertise me some channel banks, which can
operate with E1 (30 FXS)? Rack-mountable option is welcome. I've
tried to google, but I've not found nothing appropriate. More
than one E1 link is welcome too. Everyone channel banks, which
I've found, was between T1 and FXS(O), but I have Digium E1 card
and I want use for one line 30 channels and not only 24.

Thank you very much.

Sincerely
Jan Marek
--
Ing. Jan Marek               | Nez mi poslete prilohu .doc, .xls
University of South Bohemia  | nebo .ppt, prectete si, prosim,
Academic Computer Centre     | WWW stranku uvedenou na poslednim
Phone: +420-38-9032080       | radku signatury...
http://www.gnu.org/philosophy/no-word-attachments.cs.html

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Re: [asterisk-users] Round robin behavior for dialing SIP trunks...

http://www.freeswitch.org/asterisk_stuff/app_distributor.c

/b

On Aug 30, 2007, at 7:38 PM, Paul Hales wrote:

>
> We found the 'random' dialplan function worked quite well for
> something
> similar a while ago.
>
> PaulH
>
> On Thu, 2007-08-30 at 17:38 -0500, Carlos Chavez wrote:
>> I was wondering if anyone has an easy way to emulate dialing in a
>> round
>> robin fashion like when you use Zap/r1 for Zap trunks. At the moment
>> what I do is simply make a macro that will dial the sip trunks in
>> order
>> so if the first one fails it goes to the second and so on. The
>> problem
>> with this approach is that the first few SIP trunks will always be
>> busy
>> because of outgoing traffic. Is there an easy way to randomize the
>> trunks? I am guessing this will only be possible using AGI?
>>
>> _______________________________________________
>> --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
>
>
> _______________________________________________
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>

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Re: [asterisk-users] Round robin behavior for dialing SIP trunks...

We found the 'random' dialplan function worked quite well for something
similar a while ago.

PaulH

On Thu, 2007-08-30 at 17:38 -0500, Carlos Chavez wrote:
> I was wondering if anyone has an easy way to emulate dialing in a round
> robin fashion like when you use Zap/r1 for Zap trunks. At the moment
> what I do is simply make a macro that will dial the sip trunks in order
> so if the first one fails it goes to the second and so on. The problem
> with this approach is that the first few SIP trunks will always be busy
> because of outgoing traffic. Is there an easy way to randomize the
> trunks? I am guessing this will only be possible using AGI?
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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Re: [asterisk-users] where is 1.4.12?

It depends. I won't upgrade to the latest released Cisco IOS either unless
I need the added features and only after it's been out for several months.

-----Original Message-----
From: Stephen Bosch [mailto:posting@vodacomm.ca]
Sent: Thursday, August 30, 2007 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] where is 1.4.12?

shadowym wrote:
> Then you should probably use a commercial application like the Business
> Edition. I've found that once I decide to go down the open source road
it's
> a different ball game. Test with the latest and greatest release that has
> the features you need. If it's a fairly new release chances are it's not
> quite ready for prime time. Open source it not the place to be bleeding
or
> even leading edge and expect a smooth ride.

And closed source is?

-Stephen-

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Re: [asterisk-users] Hierarchical Config file (re)writing (bug 8684)

On Thu, 2007-08-30 at 16:35 -0400, Jared Smith wrote:
> On Thu, 2007-08-30 at 14:13 -0600, Steve Murphy wrote:
> > 2. Blank lines between entries will get dropped. Sorry. If you really
> > like
> > blank lines, then include a comment of a blank line.
>
> Ouch... this makes it quite cumbersome. In fact, that's the number one
> complaint I get from students in the bootcamp classes is that the
> Asterisk GUI eats their blank lines.

I'll try to see if I can put in some logic to generate line feeds based
on the line numbers from the initial read. It's not so pretty, as
comments can contain linefeeds, and there can be multiple comments
stored on the object. I have a gut feeling that preserving blank lines
will be possible, but ugly coding.

I suggest that you file this request as a bug against the new code;
that way, I won't forget. Make sure it gets assigned to me.

> > 4. Trailing comments will get dropped. Put your comments in front of a
> > variable definition, or in front of a category definition. Comments with
> > no variable decls or category decls (you know, the [general] type
> > stuff...) will not be attached to anything, and will be lost.
>
> Ouch... this one hurts too... most everyone I know uses trailing
> comments. Is there a technical reason why this has to be?
>

<caution:boring-details>
Well, "objects" in config files are basically category definitions (like
"[general]") and variable declarations (like "some-option=45"); These
little puppies are the meat inside a config file. Each one (category,
variable) ends up in a structure forming a tree of linked lists. Kinda
like a weeping willow tree, in *my* mind.

I inserted code to vacuum up all the comments as the parser progresses
thru the config file. Comments come in two flavors; first are comments
that occur on lines without any decls. And then, there are comments that
occur on the same line as the object, but after the 'object'. When we
find an object, I attach all the comments I've sucked up until then.

Trailing comments get sucked up also; there is no object to tie them to,
so they are lost.

I guess I could make up some imaginary object to tie them to; but that
complicates both the parser and possibly the routines that traverse the
category/variable tree. OR, I could add a third class of comments,
trailing comments, as another field in the objects, and attach them to
the last object scanned in the file, if we reach the end, and have some
comments in the buffers, and no object to tie them to. This also
complicates things, but not as much as the other option. I have to keep
track of the last object encountered, and stuff like that. Shouldn't be
too ugly.

So, if you actually want me to dive back into this code, and make this
enhancement, open yet one more bug and file this heart-felt wish, and
make sure it gets assigned to me, and I'll see if I can attack it.

</caution>

> Don't get me wrong, I'm happy to see the changes you've done. I'd just
> like to make this as user-friendly as possible. One of the overriding
> goals of the Asterisk GUI is to leave the config files hand-editable,
> and removing comments and blank lines makes that harder.
>

Digium pays me to make the user community happy (among other things).
So, it behooves me to acquiesce to your request. (I can hear the sound
of dictionaries being consulted!) ;)

Re: [asterisk-users] G729 copy protection

Bruce McAlister wrote:
> Bruce McAlister wrote:
>> Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to
>> initialize G.729 copy protection!
>>
>
> Hi,
>
> Could anyone from Digium please shed some light on the build
> environment for the solaris 10 g729 codec?
>
> Was it build on Solaris or OpenSolaris?
> Are there any specific versions of libraries required?
>
> I'm still having this issue, and still cannot get the codec working.
> I've had a few tips/pointer from Joe at Solaris VoIP, but now we need to
> know a little more about the build environment to see if we can actually
> get this codec working. i have tried to run the codec with asterisk
> 1.2.17, 1.2.20. 1.2.24, 1.4.4, 1.4.10, 1.4.10.1 and 1.4.11, they all
> fail with the same messages. Asteris has been built on Solaris 10 Update
> 3 patched up as of friday last week. Our focaus now is to try and get
> the codec working with asterisk 1.4.x on Solaris 10. I've also tried
> i386, i586 -> pentium4 32bit, opteron 32bit, on physical Opteron 285's
> and intel Xeon (Nacona's), all faile with the same message. The codec
> version is v32. This message comes up whether I have a valis g729
> license from Digium or not, I have tried both. In either case, I would
> assume that codec would at least load, and a "show g729" at the cli
> would work with and without a license.
>
> Has anyone been able to test this codec with asterisk?
>
> Any tips/suggestions would be greatly appreciated.
>
> Thanks
> Bruce
>

Bruce,
Please see my response to some of these questions on July 23rd.

http://lists.digium.com/pipermail/asterisk-users/2007-July/192473.html

I'm not entirely certain of what libraries we statically link in, but if you
see any problems with the output of `ldd codec_g729.so`, those will of course
need to be installed.

--
Jason Parker
Digium

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[asterisk-users] Round robin behavior for dialing SIP trunks...

I was wondering if anyone has an easy way to emulate dialing in a round
robin fashion like when you use Zap/r1 for Zap trunks. At the moment
what I do is simply make a macro that will dial the sip trunks in order
so if the first one fails it goes to the second and so on. The problem
with this approach is that the first few SIP trunks will always be busy
because of outgoing traffic. Is there an easy way to randomize the
trunks? I am guessing this will only be possible using AGI?

--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001

Re: [asterisk-users] How to handle "+" prefix

----- Original Message -----
From: "Adrian Marsh" <Adrian.Marsh@ubiquisys.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Thursday, August 30, 2007 2:34 PM
Subject: [asterisk-users] How to handle "+" prefix

> Hi,
>
> How can I have A*k convert a call from +441793xxxxxx to Dial
> 00441793xxxxxx instead?
> With the "_+." Below I can "catch" the call, but EXTEN doesn't get set
> as expected.. and then I need to figure out how to pass the call onto
> the outgoing-pstn context. Not sure if a Goto would work here...
>
>
> [outgoing-pstn-international]
> exten => _+.,1,Set(EXTEN=00${EXTEN:+1})
> exten => _+.,2,NoOp(test line: ${EXTEN})
>
> [outgoing-pstn]
> exten =>
> _0.,1,ExecIF($["${RECORDOUTBOUND}"="TRUE"],Monitor,wav|${TIMESTAMP}-${CA
> LLERID(num)}-${EXTEN}-${UNIQUEID}.WAV)
> ;exten => _0.,2,Set(CALLERID(number)=${PSTN_GLOBAL}${CALLERID(num):-2})
> exten =>
> _0.,2,ExecIF($[${LEN(${CALLERID(num)})}=4],Set,CALLERID(number)=${PSTN_G
> LOBAL}${CALLERID(num):-2})
> exten => _0.,3,Dial(${TRUNK}/${EXTEN},,W)
> exten => _0.,4,Congestion
> exten => _0.,5,Goto(outgoing-zen-out-0,${EXTEN},1)
> exten => _0.,103,Busy
> exten => _0.,104,Goto(outgoing-zen-out-0,${EXTEN},1)
>
> Adrian Marsh

Why not just use Exten => _+.,1,Goto(011${EXTEN:1})
Exten => _011.,1,Dial(......

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Re: [asterisk-users] Polycom behind NAT won't register to * server behind ALG

----- Original Message -----
From: "Eric "ManxPower" Wieling" <eric@fnords.org>
To: <asterisk@voip-pbx.ca>; "Asterisk Users Mailing List - Non-Commercial
Discussion" <asterisk-users@lists.digium.com>
Sent: Wednesday, August 22, 2007 4:08 PM
Subject: Re: [asterisk-users] Polycom behind NAT won't register to * server
behind ALG

> Henry L.Coleman wrote:
>> I think what Alex was trying to say was that Polycom IP Phones are an
>> example of immature product development. While they look very nice and
>> have a nice display the product doesn't compete very well compared to
>> other manufacturers.
>> The two most obvious flaws are that they cannot be NAT'ed so they cannot
>> be used as Off Premise eXtensions phones and the other being that they
>> take so long to configure and re-boot. I have a golden rule with any
>> phone
>> that I plan on installing for a customer....If I can't get it working
>> within 20 minutes then don't use it. I'm afraid Polycom breaks my golden
>> rule.
>> With such a lot of competition in this market they should have sorted
>> this
>> out two years ago.
>>
>
> Reboots should not happen very often. This is a non-issue for most
> people.
>
> I've never seen a phone that could not work with NAT with Asterisk.
> Polycoms work just fine with NAT and Asterisk. The nice thing about
> Asterisk's NAT support is that the phone does not need to support NAT.

Eric,
Try using 5 Polycom's at a remote location behind NAT. Let me know when you
need a drink ;) . I had a client with such an issue and the "fix" was a nice
Edgemark or Sonicwall firewall that are set up for SIP and NAT issues. I
prefer the Sonicwall.

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Re: [asterisk-users] OT: DELL Platforms

----- Original Message -----
From: "Steve Totaro" <stotaro@first-notification.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Tuesday, August 28, 2007 11:51 AM
Subject: Re: [asterisk-users] OT: DELL Platforms

> Dovid B wrote:
>> <snip>
>>
>>> I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM
>>> and a couple 250gig SATA drives. Totally VoIP so I cannot comment on
>>> cards or interrupts, but so far it has been flawless.
>>>
>>> I would like to see how many G729/ULAW conversions it could handle. How
>>> would I go about benchmarking that?
>>>
>>> Thanks,
>>> Steve
>>>
>> </snip>
>>
>> I would advise against the SC Series because there is no RAC card option.
>> You never know that you needed one till you need to format a box from
>> your
>> hotel room ;)
>>
>> I have used some Poweredge 1850's with Asterisk (VOIP Only) and I have
>> been
>> real happy.
>>
>>
>
> Not so much a problem since the CoLo is staffed 24/7 and glad to help.
> Soon to add KVMoIP.
>
> There is a rudementary (have not played with it yet)
> /The latest industry-standard Intelligent Platform Management Interface
> (IPMI) 2.0 Baseboard Management Controller (BMC) allows remote,
> out-of-band management over a network or serial connection with any
> industry-standard IPMI management program.
>
> /I will play with this feature a bit before investing in KVMoIP.
>
> Thanks,
> Steve Totaro

Steve,
You have a point but you can't cycle the power with KVM over VOIP. The DC is
staffed 24/7 but it takes time till they pick up, reboot the server etc.
Also who wants to pay them to format a box if I can do it remotely ;)

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Re: [asterisk-users] AsteriskNOW Web GUI

I am a long time user and reseller of Thirdlane PBX Manager. From my
standpoint the implementation tools are outstanding and the fact that
the files are easy to follow means it allows a consultant to comstomize
the behavior yet allow the end user to maintain going forward.

good luck!

Steve Totaro wrote:
> The README is here: svn co
> http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
>
> /Configuration
> =============
> You may install sample configuration files by doing "make samples".
> Also you
> will need to edit your Asterisk configuration files to enable the GUI
> properly,
> specifically:
>
> 1) In http.conf:
>
> [general]
> enabled = yes
> enablestatic = yes/
>
> I am looking at Thirdlane's solution now. Very impressive and modest cost.
>
> Thanks,
> Steve
>
> bkruse wrote:
>
>> As Tzafrir stated, it will NOT work with 1.2.x.
>>
>> Where is this html.conf, which README? I will update it.
>>
>> I will write a brief page on setting up the *GUI for all who want to
>> know......
>>
>> There are SOME GUI's that work with 1.2, however, I almost guarantee
>> none of them are client side, such as this one.
>>
>> -bk
>>
>>
>> Steve Totaro wrote:
>>
>>
>>> Will this work on 1.2.x? I just installed it and did make samples.
>>>
>>> The README references a file called html.conf which does not exist and
>>> also abruptly ends with the word "to" on a blank line.
>>>
>>> Besides that, what would the URL be for AsteriskNow? Is that
>>> customizable in the elusive html.conf file?
>>>
>>> Any GUIs that are easily installed on existing systems and work with 1.2.x?
>>>
>>> Thanks,
>>> Steve
>>>
>>> bkruse wrote:
>>>
>>>
>>>
>>>> svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow

>>>> thegui; cd thegui; sh configure; make && sudo make install ; clear ;
>>>> echo 'completed'
>>>>
>>>> -bk
>>>> Yann JOUANIN wrote:
>>>>
>>>>
>>>>
>>>>
>>>>> You can do it from svn server , I think there is a page in the wiki
>>>>>
>>>>>
>>>>>
>>>>> Best,
>>>>>
>>>>>
>>>>>
>>>>> yann
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> *De :* asterisk-users-bounces@lists.digium.com
>>>>> [mailto:asterisk-users-bounces@lists.digium.com] *De la part de*
>>>>> Jeremy Mann
>>>>> *Envoyé :* vendredi 24 août 2007 17:30
>>>>> *À :* Asterisk Users Mailing List - Non-Commercial Discussion
>>>>> *Objet :* [asterisk-users] AsteriskNOW Web GUI
>>>>>
>>>>>
>>>>>
>>>>> Is the web GUI for AsteriskNOW able to be loaded on an existing
>>>>> server(that was installed from ubuntu-server and asterisk loaded from
>>>>> source)?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>
>>>
>
>
>
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Re: [asterisk-users] Redundancy / Failover

<snip>
> question2: it's possible read registration data from astdb from python/php
> (or it is possible write sip registrations to mysql/sqlite? i do not
> want realtime because of NAT issues)
</snip>

Marek,
What NAT issues can realtime create that there won't be in static ?

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Re: [asterisk-users] Redundancy / Failover

You may want to consider upgrading your version of asterisk. Next you can try using SER + Asterisk + Heartbeat.

----- Original Message -----

From: Khaled Chehab

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Cc: asterisk-users-bounces@lists.digium.com

Sent: Tuesday, August 21, 2007 3:05 PM

Subject: [asterisk-users] Redundancy / Failover

 

Dears

 

Any one succeeded to make Redundancy / Failover  with  asterisk 1.4.9 on centos with kernel 2.6.9-55.EL.   

Can you please send me the documentation link on how to or write down how to .

 

 

 

Regards

 

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Re: [asterisk-users] Queue Agents on Remote Asterisk server?

How about sending a SipHeader to the second box and then on the second box look for the header. If the header does not exist then ring the extension normally. If the header is there than send back congestion (basically have a gotoif before it hits the Exten => Foo,1,Voicemail)

----- Original Message -----

From: Aubrey Wells

To: asterisk-users@lists.digium.com

Sent: Wednesday, August 29, 2007 8:25 PM

Subject: [asterisk-users] Queue Agents on Remote Asterisk server?

Hi,

I have a main Asterisk server, and a server at a branch location connected via a IAX2 trunk. I want to have a queue at the main location that has people from both locations as members. I got this working, but the trouble comes when the round-robin logic selects a member at the branch office to call. If that user is unavailable, their voicemail answers the call, and the main server detects this as an answered call and assumes the agent answered. This is obviously not what I want, as I would like for the call to roll to one of the other agents. Has anyone come across this before? Solutions?

Thanks!

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

Aubrey Wells

---

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Re: [asterisk-users] problem with rfc2833

Are you using 1.4.X on one and 1.2.X on another ?

----- Original Message -----
From: "Jerry Geis" <geisj@pagestation.com>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, August 02, 2007 3:34 PM
Subject: [asterisk-users] problem with rfc2833

>I have the following:
>
> pri box incoming/outgoing on box 1 connected through SIP to box 2.
> The box 1 to box 2 has dtmfmode=rfc2833.
> With this setting calls going out of box2 through box 1 the sendDTMF()
> mode does not do anything.
>
> When I change dtmfmode=info I at least hear the sendDTMF() digits.
>
> Why doesnt rfc2833 work?
>
> Jerry
>
>
> _______________________________________________
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Re: [asterisk-users] where is 1.4.12?

----- Original Message -----
From: "Jared Smith" <jsmith@digium.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Thursday, August 30, 2007 5:41 PM
Subject: Re: [asterisk-users] where is 1.4.12?

> On Thu, 2007-08-30 at 08:02 -0500, Eric "ManxPower" Wieling wrote:
>> As I understand it, Digium does NO formal QA testing before the free
>> Asterisk/Zaptel/libPRI releases. Asterisk Business Edition is a
>> different story and gets extensive QA testing.
>
> As I understand it, that's simply due to a lack of resources. At the
> Asterisk Developer's Conference earlier this year, the Asterisk
> Developers were all pretty much in agreement that more needed to be done
> in this area, but that it would have to be a combined effort between the
> Asterisk community and Digium, as Digium simply doesn't have the
> resources at this point to do it all itself.
>
>> On IRC I have been a "vocal user from hell" about the QA issues of
>> Digium open source products.
>
> I've tried to be vocal about this too. And now that I'm working for
> Digium, I'd be happy to try to coordinate an effort between the
> community and Digium to try to come up with a framework where we can all
> work together to make this happen.
>
>

Jared,
I do not have much time on my hands but I do have some spare servers up and
running. If there are ways for us "end users" to help I will gladly give
back to those that gave to me.

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[asterisk-users] Digium Asterisk Appliance reviews?

Hi All -

Has anyone had a chance to use the Asterisk Appliance yet? Any
thoughts or reactions? I have a couple of clients waiting on the
Zaptel version, but maybe somebody has used the VoIP-only version?

Thanks,
Noah

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Re: [asterisk-users] How long to detect an "h" exten?

Can you explain this question?
Just to clearify, exten => h will execute as soon as Asterisk is aware
that the channel was hung up. While app_hangup will execute a hangup
on an active channel.

On 8/30/07, Gavin Henry <gavin.henry@gmail.com> wrote:
> Dear All,
>
> How long should it take before a exten => h,1,Hangup() kicks in,
> versus a exten => s,n,Hangup()
>
> I'm just about to test, but thought I'd ask.
>
> --
> http://www.suretecsystems.com/services/openldap/
>
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Re: [asterisk-users] FYI

Indeed very interesting and informative. I think this has been covered
in past issues of 2600, but this is the first time these docs are
available.

Thank you

On 8/30/07, Joe Acquisto <joea@j4computers.com> wrote:
> http://www.wired.com/print/politics/security/news/2007/08/wiretap
>
>
>
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Re: [asterisk-users] Problems with overlap dial and Xorcom Astribank BRI

For anyone who is interested in the solution:

It seems Asterisk detected a busy signal. Setting 'busydetect=no' in
zapata.conf solved this problem.

Lars

--
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Re: [asterisk-users] Testing Framework

Matt Riddell wrote:
> Should these tests be added to Asterisk-Addons or maintained outside of
> the tree?

If people start writing test utilities, I would be happy to host them in a
subversion repository. Depending on the size of this stuff, it could probably
go into the main Asterisk repository. We'll see where things go ...

--
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] Hierarchical Config file (re)writing (bug 8684)

On Thu, 2007-08-30 at 14:13 -0600, Steve Murphy wrote:
> 2. Blank lines between entries will get dropped. Sorry. If you really
> like
> blank lines, then include a comment of a blank line.

Ouch... this makes it quite cumbersome. In fact, that's the number one
complaint I get from students in the bootcamp classes is that the
Asterisk GUI eats their blank lines.

> 4. Trailing comments will get dropped. Put your comments in front of a
> variable definition, or in front of a category definition. Comments with
> no variable decls or category decls (you know, the [general] type
> stuff...) will not be attached to anything, and will be lost.

Ouch... this one hurts too... most everyone I know uses trailing
comments. Is there a technical reason why this has to be?

Don't get me wrong, I'm happy to see the changes you've done. I'd just
like to make this as user-friendly as possible. One of the overriding
goals of the Asterisk GUI is to leave the config files hand-editable,
and removing comments and blank lines makes that harder.

--
Jared Smith
Community Relations Manager
Digium, Inc.

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[asterisk-users] Hierarchical Config file (re)writing (bug 8684)

Anyone remember the problem with writing out config files that had
#include directives in them? You'd get a single, flat config file when
you saved it back out. I have heard a few howls of complaint! (see bug
8684)

Well, I just checked in a fix for this into trunk. My simple tests say
it's working, but the "real thing" may say otherwise. It will re-write
the config file in question, and also re-write all the files included by
it. All will have the little header inserted at the top, revealing that
the file is generated. I only see a few places in the code where the
text_file_save function is called: in the manager (AMI) code, and the
voicemail app(-- if you are in realtime, I think.)

What might be different:

1. If you included the same file twice (or more), you will see that,
after it is rewritten to disk, now two different files are included (one
a copy of the first), and if changes are made to one, and not the other,
then they may not be the same anymore.

2. Blank lines between entries will get dropped. Sorry. If you really
like
blank lines, then include a comment of a blank line.

3. Comments preceded by a large amount of whitespace will probably have
most of the preceding space dropped. sorry. Start your or comment at the
beginning of a line, or immediately after statements, and you'll keep
the spaces that way.

4. Trailing comments will get dropped. Put your comments in front of a
variable definition, or in front of a category definition. Comments with
no variable decls or category decls (you know, the [general] type
stuff...) will not be attached to anything, and will be lost.

5. Right now, #exec lines will be written out, if they are included, but
their output will not be included in the saved set. (Well, you could
look in /var/tmp, but they aren't ref'd by anything.). But, I think you
just want #exec's to get repeated as you placed them. Their output is a
dynamic thing anyway...!

So, I ask those who were interested in seeing this kind of
functionality, to test it out in trunk and report the bugs via the bug
tracker. I know there are some who would like this in 1.4, and after we
toughen up the code in trunk, maybe we could call it a bugfix and get it
in 1.4, if that is the right thing to do.

murf

--
Steve Murphy
Software Developer
Digium

[asterisk-users] Testing Framework

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

So, now that we've all complained about the state of testing of Open
Source versions of Asterisk, lets do something about it.

I propose we start with a list of things that we think should be tested
in Asterisk, and means to test them.

Maybe we could run certain tests based on the changes between minor
versions?

Anyway lets start.

Call Volumes

1) Call volume up to x channels from SIP to SIP (i.e. sipp)
2) Call volume up to x channels from IAX2 to SIP
3) Call volume up to x channels from IAX2 to IAX2

Application testing

4) Connect x calls between techs to Meetme (leave running for 1 hour)
5) Connect x concurrent calls to VoiceMail

Call Centre Testing

6) Send x calls to a queue with no agents in it, leave them holding for
x minutes
7) Run x calls against AMD connected to recorded known good files

Recording

8) Run x calls recording simultaneously from an automatically generated
call, play ulaw/alaw - compare outputs.

You get the idea.

If people can add to this list, I can start making a few scripts and
programs that will test them (as I'm sure others can).

If we end up with a complete list, I'm sure some of our individual QA
departments can take the responsibility for certain items.

The call volume ones are obviously going to either need a live person to
dial in at volume and check everything is ok, or a recording which can
later be checked.

I'm of the opinion that the majority of tests should test individual
components, but that we should also form some "Application Type"
frameworks so that we can test integration between Asterisk apps.

Any takers? Add to the list? If there is something you believe is
mission critical to your business, write up a test case for it, and
we'll all try to code something that can run automatically to test it.

If we try and keep to ANSI C for the testing apps, Digium should be able
to run them on their multi platform machines as well.

Should these tests be added to Asterisk-Addons or maintained outside of
the tree?

Anyway, what do you think? Feasible? I already have a few tests here and
I'm sure others have a few too. Lets put them all together and get a
framework going.

- --
Kind Regards,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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