Friday, November 30, 2007

Re: [asterisk-users] REFER mesage extraction using SIP_HEADER

On Dec 1, 2007, at 12:30 AM, asterisk-users-request@lists.digium.com
wrote:

> I would like to extract the information present in the SIP REFER
> message that comes to asterisk. Would SIP_HEADER() allow me to do that
> ? I have used SIP_HEADER() for extracting the to and from SIP headers
> previously.

I wanted to do the exact same thing a while ago. However it is not
possible as far as I can tell.

I've tried it and verified the headers are being sent, but asterisk
can't see them. It can read the headers from the original INVITE.

This bug report:
http://bugs.digium.com/view.php?id=4934

Complained of the same thing, but was ended as too much work and
folks weren't sure it's even correct.

Then this one:
http://bugs.digium.com/print_bug_page.php?bug_id=8378

Talks about the Refered-By header in REFER messages, which seems to
have been folded in to 1.4. It didn't solve the general case of other
headers, however.

I worked around it in my case, since the original invite actually had
what I needed most of the time. Some odd cases just will remain broken.

Norman Franke
ASD, Inc.
www.myasd.com

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Off-Topic: Avaya

Salvatore Giudice wrote:
> They are cheap. You only have to pay for the box and the
> maintenance percentage.

That is indeed the Avaya way. First you buy it, then you rent it. Stop
paying their maintenance fees and their dial into your PBX and cripple
the OS by removing customer maintenance command permissions.

> Hell, Avaya won't even
> give you root on any of their servers. You cant audit the box and you can't
> poll them unless you pay them money to join their partner program and get
> their SDK. If you already have Avaya, you should just buy Message Networking
> or a Mitel voicemail server if you want seamless voicemail with Avaya.
>
> However, you should know that using Avaya is probably a bad idea to begin
> with. Until February 07, the majority Avaya's soft switch products were
> running on Redhat 9, which was unsupported since 2003. Avaya was only
> managing a dozen packages and they've always left it up to the customer to
> know when they need an update, requiring the customer to request a field
> load. It has to be the worst update model in the industry when it comes to
> infrastructure monitoring and patching. By using Avaya, you are blindly
> trusting them to properly maintain a Linux appliance. This is something they
> are not capable of and you can't even audit them.
>
> Avaya is what happens to organizations when they have ignorant telecom
> infrastructure engineers deciding what products to buy. Avaya focuses sales
> on those engineers because they k now their products won't pass
> certification by network, systems, or security engineers. Telecom engineers
> only look for features and usually get their asses handed to them after they
> put Avaya VoIP into their infrastructure.
>

Bravo. A well-deserved lambasting of this awful vendor.

--
# Jesse Molina
# Mail = jesse@opendreams.net
# Page = page-jesse@opendreams.net
# Cell = 1.602.323.7608
# Web = http://www.opendreams.net/jesse/

_______________________________________________
--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

Posted by at No comments:

[asterisk-users] Asterisk & Cisco calling Name

Anyone see an issue on asterisk 1.2 that it will not accept the invite
from a Cisco gateway. If I turn off voice service voip signaling
forward unconditional then Asterisk accepts the call but without cname.
Below is a trace.

Any help is appreciated.

Thanks

John Bittner
Simlab.net


voippbx01*CLI>
<-- SIP read from 216.86.35.24:63549:
INVITE sip:9734333001@69.60.198.130:5060 SIP/2.0
Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56
From: <sip:9733901090@216.86.35.24>;tag=4F9EF08-163B
To: <sip:9734333001@69.60.198.130>
Date: Sat, 01 Dec 2007 05:23:25 GMT
Call-ID: 602E8F94-9F0411DC-8ACEEC29-3723F693@216.86.35.24
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 1613584196-2667844060-2152857615-892193345
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: "pending" <sip:9733901090@216.86.35.24>;party=calling;screen=yes;privacy=off
Timestamp: 1196486605
Contact: <sip:9733901090@216.86.35.24:5060>
Expires: 180
Allow-Events: telephone-event
MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=uniqueBoundary
Content-Length: 680

--uniqueBoundary
Content-Type: application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 6852 2375 IN IP4 216.86.35.24
s=SIP Call
c=IN IP4 216.86.35.24
t=0 0
m=audio 18472 RTP/AVP 0 101
c=IN IP4 216.86.35.24
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional

IAM,
PRN,isdn*,,NI***,
USI,rate,c,s,c,1
USI,lay1,ulaw
TMR,00
CPN,04,,1,9734333001
CGN,04,,1,y,4,9733901090
CPC,09
FCI,,,,,,,y,
GCI,602d57449f0411dc8052000f352dca41
UFC,GEN,5,gentf,79
UFC,GEN,5,fachd,9f8b0100
UFC,GEN,5,inpdu,02010106072a8648ce150004

--uniqueBoundary--

--- (21 headers 33 lines)---
Using INVITE request as basis request - 602E8F94-9F0411DC-8ACEEC29-3723F693@216.86.35.24
Sending to 216.86.35.24 : 5060 (non-NAT)
Found peer '216.86.35.24'
Transmitting (no NAT) to 216.86.35.24:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56;received=216.86.35.24
From: <sip:9733901090@216.86.35.24>;tag=4F9EF08-163B
To: <sip:9734333001@69.60.198.130>;tag=as39c359be
Call-ID: 602E8F94-9F0411DC-8ACEEC29-3723F693@216.86.35.24
CSeq: 101 INVITE
User-Agent: SimlabVOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9734333001@69.60.198.130>
Content-Length: 0


---
Destroying call '602E8F94-9F0411DC-8ACEEC29-3723F693@216.86.35.24'
voippbx01*CLI>
<-- SIP read from 216.86.35.24:5060:
ACK sip:9734333001@69.60.198.130:5060 SIP/2.0
Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56
From: <sip:9733901090@216.86.35.24>;tag=4F9EF08-163B
To: <sip:9734333001@69.60.198.130>;tag=as39c359be
Date: Sat, 01 Dec 2007 05:23:25 GMT
Call-ID: 602E8F94-9F0411DC-8ACEEC29-3723F693@216.86.35.24
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

Hi,
I am using the OS which bundled with AsteriskNow
 
 
----- Original Message -----
To: Newbie
Sent: Saturday, December 01, 2007 12:25 PM
Subject: Re: [asterisk-users] Registration state: Failed

Hmmm, what OS you are using,,,this could be related to "Access Control Lists"..but i guess that is in Solaris  

On 11/30/07, Newbie <newbie@pbxsoftwares.com> wrote:
Hello,
 
After I turned on "full=>" in logged.conf .. I got the following:
 
[Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 <sip:998@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:998@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:998@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
any idea or clue?
Thanks a lot in advance
Regards
Winanjaya
 
----- Original Message -----
To: Newbie
Sent: Saturday, December 01, 2007 11:50 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory.
 
Thanks,
 
Vivek 

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
 
Line 1:
Registration Status: Failed
 
PSTN Line 1:
Registration Status: Failed
 
I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow..
 
since I am very new with this.. I don't know why this problem occurs ... could any body please help?
 
Thanks & Regards
Winanjaya
 
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=h263
allow=h263p
register=998:1234@line1/998
register=999:1234@pstnline1/999
[line1]
type=peer
host=dynamic
defaultip=172.16.1.74
fromuser=998
secret=1234
fromdomain=172.16.1.169
 
[line2]
type=peer
host=dynamic
defaultip=172.16.1.74
username=999
secret=1234
fromdomain=172.16.1.169
 
Command> sip show peers  Name/username              Host            Dyn Nat ACL Port     Status                pstnline1/999              (Unspecified)    D          0        Unmonitored            line1                      (Unspecified)    D          0        Unmonitored            250/250                    172.16.1.88      D          27778    Unmonitored            2500                       (Unspecified)    D          0        Unmonitored            251                        (Unspecified)    D          0        Unmonitored            5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] 
 
 

 
 
----- Original Message -----
Sent: Saturday, December 01, 2007 11:34 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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



Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

you can also look at this...
 
 
"I has this error initially with Asterisk server when I try to register.

" Device does not match ACL "

got it resolved by setting Caller ID Name : " users exten "


 
On 11/30/07, Vivek Shrivastava <vivshrivastava@gmail.com> wrote:
Hmmm, what OS you are using,,,this could be related to "Access Control Lists"..but i guess that is in Solaris  


On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Hello,
 
After I turned on "full=>" in logged.conf .. I got the following:
 
[Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 <sip:998@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:998@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:998@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
any idea or clue?
Thanks a lot in advance
Regards
Winanjaya
 
----- Original Message -----
To: Newbie
Sent: Saturday, December 01, 2007 11:50 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory.
 
Thanks,
 
Vivek 

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
 
Line 1:
Registration Status: Failed
 
PSTN Line 1:
Registration Status: Failed
 
I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow..
 
since I am very new with this.. I don't know why this problem occurs ... could any body please help?
 
Thanks & Regards
Winanjaya
 
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=h263
allow=h263p
register=998:1234@line1/998
register=999:1234@pstnline1/999
[line1]
type=peer
host=dynamic
defaultip=172.16.1.74
fromuser=998
secret=1234
fromdomain=172.16.1.169
 
[line2]
type=peer
host=dynamic
defaultip=172.16.1.74
username=999
secret=1234
fromdomain=172.16.1.169
 
Command> sip show peers  Name/username              Host            Dyn Nat ACL Port     Status                pstnline1/999              (Unspecified)    D          0        Unmonitored            line1                      (Unspecified)    D          0        Unmonitored            250/250                    172.16.1.88      D          27778    Unmonitored            2500                       (Unspecified)    D          0        Unmonitored            251                        (Unspecified)    D          0        Unmonitored            5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] 
 
 

 
 
----- Original Message -----
Sent: Saturday, December 01, 2007 11:34 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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




Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

Hmmm, what OS you are using,,,this could be related to "Access Control Lists"..but i guess that is in Solaris  

On 11/30/07, Newbie <newbie@pbxsoftwares.com> wrote:
Hello,
 
After I turned on "full=>" in logged.conf .. I got the following:
 
[Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 <sip:998@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:998@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:998@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:999@172.16.1.74>' failed for ' 172.16.1.169' - Device does not match ACL
any idea or clue?
Thanks a lot in advance
Regards
Winanjaya
 
----- Original Message -----
To: Newbie
Sent: Saturday, December 01, 2007 11:50 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory.
 
Thanks,
 
Vivek 

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
 
Line 1:
Registration Status: Failed
 
PSTN Line 1:
Registration Status: Failed
 
I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow..
 
since I am very new with this.. I don't know why this problem occurs ... could any body please help?
 
Thanks & Regards
Winanjaya
 
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=h263
allow=h263p
register=998:1234@line1/998
register=999:1234@pstnline1/999
[line1]
type=peer
host=dynamic
defaultip=172.16.1.74
fromuser=998
secret=1234
fromdomain=172.16.1.169
 
[line2]
type=peer
host=dynamic
defaultip=172.16.1.74
username=999
secret=1234
fromdomain=172.16.1.169
 
Command> sip show peers  Name/username              Host            Dyn Nat ACL Port     Status                pstnline1/999              (Unspecified)    D          0        Unmonitored            line1                      (Unspecified)    D          0        Unmonitored            250/250                    172.16.1.88      D          27778    Unmonitored            2500                       (Unspecified)    D          0        Unmonitored            251                        (Unspecified)    D          0        Unmonitored            5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] 
 
 

 
 
----- Original Message -----
Sent: Saturday, December 01, 2007 11:34 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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



Posted by at No comments:

[asterisk-users] REFER mesage extraction using SIP_HEADER

Hi * users,

I would like to extract the information present in the SIP REFER
message that comes to asterisk. Would SIP_HEADER() allow me to do that
? I have used SIP_HEADER() for extracting the to and from SIP headers
previously.

Thanks

Regards
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University

Tel: 1-646-387-5998

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

Hello,
 
After I turned on "full=>" in logged.conf .. I got the following:
 
[Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 <sip:998@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 <sip:998@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 <sip:998@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 <sip:999@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 <sip:999@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 <sip:999@172.16.1.74>' failed for '172.16.1.169' - Device does not match ACL
any idea or clue?
Thanks a lot in advance
Regards
Winanjaya
 
----- Original Message -----
To: Newbie
Sent: Saturday, December 01, 2007 11:50 AM
Subject: Re: [asterisk-users] Registration state: Failed

well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory.
 
Thanks,
 
Vivek 

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com> wrote:
Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
 
Line 1:
Registration Status: Failed
 
PSTN Line 1:
Registration Status: Failed
 
I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow..
 
since I am very new with this.. I don't know why this problem occurs ... could any body please help?
 
Thanks & Regards
Winanjaya
 
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=h263
allow=h263p
register=998:1234@line1/998
register=999:1234@pstnline1/999
[line1]
type=peer
host=dynamic
defaultip=172.16.1.74
fromuser=998
secret=1234
fromdomain=172.16.1.169
 
[line2]
type=peer
host=dynamic
defaultip=172.16.1.74
username=999
secret=1234
fromdomain=172.16.1.169
 
Command> sip show peers  Name/username              Host            Dyn Nat ACL Port     Status                pstnline1/999              (Unspecified)    D          0        Unmonitored            line1                      (Unspecified)    D          0        Unmonitored            250/250                    172.16.1.88      D          27778    Unmonitored            2500                       (Unspecified)    D          0        Unmonitored            251                        (Unspecified)    D          0        Unmonitored            5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] 
 
 

 
 
----- Original Message -----
Sent: Saturday, December 01, 2007 11:34 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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


Posted by at No comments:

Re: [asterisk-users] Shared line appearance phones?

On Nov 29, 2007 5:49 AM, Mark Wiater <mwiater@cablespeed.com> wrote:
Russell Bryant wrote:
> Ron McCarthy wrote:
>> Asterisk 1.4 im guessing? I did not know the Snom's worked with that,
>> Ill have to check it out then!
>
> The way it is implemented in Asterisk is a bit interesting.  It uses the
> existing device state support (hints, BLF) to manage the buttons for shared
> lines.  Asterisk changes the state of these virtual "shared lines" to different
> states, and the light on the phone reflects the state (in use, ringing, on hold).

I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several
Polycom IP430's.

Might you be willing to share some specific configurations for such a situation?

 
Mark,
 
That's what I have been trying very unsuccessfully to do as well.  It seems to be something that can't be done in a few minutes here and there of spare time :-)
thanks

mark


_______________________________________________
--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



--
Lacy Moore
Somewhere I wish I wasn't
Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory.
 
Thanks,
 
Vivek 

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com> wrote:
Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
 
Line 1:
Registration Status: Failed
 
PSTN Line 1:
Registration Status: Failed
 
I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow..
 
since I am very new with this.. I don't know why this problem occurs ... could any body please help?
 
Thanks & Regards
Winanjaya
 
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=h263
allow=h263p
register=998:1234@line1/998
register=999:1234@pstnline1/999
[line1]
type=peer
host=dynamic
defaultip=172.16.1.74
fromuser=998
secret=1234
fromdomain=172.16.1.169
 
[line2]
type=peer
host=dynamic
defaultip=172.16.1.74
username=999
secret=1234
fromdomain=172.16.1.169
 
Command> sip show peers  Name/username              Host            Dyn Nat ACL Port     Status                pstnline1/999              (Unspecified)    D          0        Unmonitored            line1                      (Unspecified)    D          0        Unmonitored            250/250                    172.16.1.88      D          27778    Unmonitored            2500                       (Unspecified)    D          0        Unmonitored            251                        (Unspecified)    D          0        Unmonitored            5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] 
 
 

 
 
----- Original Message -----
Sent: Saturday, December 01, 2007 11:34 AM
Subject: Re: [asterisk-users] Registration state: Failed

 
Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com > wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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


Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
 
Line 1:
Registration Status: Failed
 
PSTN Line 1:
Registration Status: Failed
 
I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow..
 
since I am very new with this.. I don't know why this problem occurs ... could any body please help?
 
Thanks & Regards
Winanjaya
 
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=h263
allow=h263p
register=998:1234@line1/998
register=999:1234@pstnline1/999
[line1]
type=peer
host=dynamic
defaultip=172.16.1.74
fromuser=998
secret=1234
fromdomain=172.16.1.169
 
[line2]
type=peer
host=dynamic
defaultip=172.16.1.74
username=999
secret=1234
fromdomain=172.16.1.169
 
Command> sip show peers  Name/username              Host            Dyn Nat ACL Port     Status                pstnline1/999              (Unspecified)    D          0        Unmonitored            line1                      (Unspecified)    D          0        Unmonitored            250/250                    172.16.1.88      D          27778    Unmonitored            2500                       (Unspecified)    D          0        Unmonitored            251                        (Unspecified)    D          0        Unmonitored            5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] 
 
 
 
----- Original Message -----
Sent: Saturday, December 01, 2007 11:34 AM
Subject: Re: [asterisk-users] Registration state: Failed

Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com> wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation

Tilghman Lesher wrote:
> On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote:
>
>> [snip]
>> The issue is that I have, per "virtual pbx" (i.e. home or business), two
>> contexts that these get used from. The "internal-xyzzy" and
>> "incoming-xyzzy" contexts (one for each pbx, ie. "xyzzy" is "home" or else
>> it's "office").
>>
>> I was wondering if there wasn't a more flexible solution to this issue,
>> than hard-coding a "Goto(default,s,1)" into them (I have no default
>> context, because it would be meaningless).
>>
>> Perhaps using "Gosub" and "Return". Or do I need to hack the macro, and
>> pass in a 3rd argument (bletch)?
>>
>
> MacroExit or Gosub/Return would certainly be possibilities.
>
> The main thing to note is that this macro that you call standard is actually
> just an arbitrary example. It is by no means perfect, so feel free to adapt
> it to your own liking.
>

Sure. I just figured that it would be nice if the canned macros worked
out-of-the-box without modification, in the real world.

I suppose I could file a bug, and then submit patches for the macro and
documentation...

-Philip


_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Registration state: Failed

Hi,
 
x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues.
 
Thanks,
 
Vivek

 
On 11/30/07, Newbie <newbie@pbxsoftwares.com> wrote:
Dear Support,
 
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line.
 
I have 3 extensions:
 
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
 
I have created 998 and 999 to the user extension list of the AsteriskNow
 
why I still got Registration state: Failed for both Line 1 status and PSTN Line status ?
 
 
my topology is below:
 
Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line
 
Please help
 
Thanks  a lot in advance
 
Regards
Winanjaya
 
 

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Asterisk on Pcengines Alix board

> Is the PCI slot large enough for full height, half length PCI boards ?  
 
Yes.
 
> Has you heard of a  PCI Express version ? 
 
No but the way chipsets are coming down in price, I would imagine someone will have it soon.
 
John 
 
  
Posted by at No comments:

[asterisk-users] How to setup redundant SIP peers

Hello list,

I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:

[peer1]
type=peer
host=10.10.10.1

[peer2]
type=peer
host=10.10.10.2

Now dialout is no problem. Extensions.conf says:

exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30)

But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
answering (in 3sec) or send "50x" error?
Next idea is to use both peers in round-robin, if they are working.

Could someone help?

Regards
Thomas

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] IAX complaints? What are they?

Daryl G. Jurbala wrote:
> How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and
> asterisk would stop accepting IAX connections in less than a day and
> would need to be restarted.

It has been a continuously worked on task (ever since a few months ago).
Russell Bryant and others have been working on it and has improved its
reliability to the point of fixing most if not all of the previously
outstanding issues. I recommend trying it again.

--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Suppressing certain queue announcement voice prompts

> > > > Short of replacing a sound file with a sound file containing only
> > > > a short period of silence, is there any way to suppress certain
> > > > sounds from playing during queue processing by configuring for
> > > > example queues.conf or other similar files?
> > >
> > > Which announcements are you trying to not play?
> >
> > queue-thankyou for instance, to name one. Or any other of the queue-*
> > files in general. From time to time it can be convenient to change
> the
> > exact prompts played (order and contents) due to language differences
> > and personal preference of the end-users.
>
> The question is more like what exactly do you mean with "from time to
> time"?
>
> Anyway, your best option is probably to create one or more prompt
> languages by copying the English prompts to a new directory like "en2",
> "en3" and then use Set(LANGUAGE=en3) in the dialplan when you think
> this is appropriate. For each of these artificial languages you can now
> decide how to modify the sound files.
>
> Cheers, Philipp

Again, very good advice thank you Philipp. And probably a very reasonable
way to do this if dynamic behaviour is needed. But in my case time-to-time
was meant as "every once in a while there is a particullar installation that
requires this". So statically doing this is ok in my case.

I'll continue with my replace-with-silence-file method for now. Thanks for
the input.

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?

looks like something wrong with the dial plan in the extensions.conf.. i would recommend start debug on and see the content of "full" log  may be that give some clue.
 
Thanks,
 
Vivek

 
On 11/30/07, Russell Brown <russell@lls.lls.com> wrote:

I have two Asterisk systems that can route to each other via a VPN with
firewalls disabled for testing purposes.

Each Server can see (tested via nmap) UDP port 5060 on the other.

So...  I thought that I could simply use a Dial command in Server A's
config to place a SIP call to Server B...  but it doesn't seem to work.

Server A (192.168.1.33) has:

       exten => *136,1,Dial(SIP/90@10.10.111.13,30)

but whenever a user on Server A dials '*136' the call doesn't complete
and the CLI shows:

       Executing [*136@from-sip:1] Dial("SIP/112-0071f650", "SIP/90@10.10.111.13|30") in new stack
       -- Called 90@10.10.111.13
       -- SIP/10.10.111.13-00793520 is circuit-busy
       == Everyone is busy/congested at this time (1:0/1/0)

I can't see anything in Server B's logs from 192.168.1.33

What am I missing?

Any pointers to help me get this working?

--
Regards,
    Russell
--------------------------------------------------------------------
| Russell Brown          | MAIL: russell@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems     | WWW Work: http://www.lls.com              |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk         |
--------------------------------------------------------------------

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Suppressing certain queue announcement voice prompts

Hi!

> > > Short of replacing a sound file with a sound file containing only a
> > > short period of silence, is there any way to suppress certain sounds
> > > from playing during queue processing by configuring for example
> > > queues.conf or other similar files?
> >
> > Which announcements are you trying to not play?
>
> queue-thankyou for instance, to name one. Or any other of the queue-* files
> in general. From time to time it can be convenient to change the exact
> prompts played (order and contents) due to language differences and personal
> preference of the end-users.

The question is more like what exactly do you mean with "from time to
time"?

Anyway, your best option is probably to create one or more prompt
languages by copying the English prompts to a new directory like "en2",
"en3" and then use Set(LANGUAGE=en3) in the dialplan when you think this
is appropriate. For each of these artificial languages you can now decide
how to modify the sound files.

Cheers, Philipp


_______________________________________________
--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

Posted by at No comments:

[asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?

I have two Asterisk systems that can route to each other via a VPN with
firewalls disabled for testing purposes.

Each Server can see (tested via nmap) UDP port 5060 on the other.

So... I thought that I could simply use a Dial command in Server A's
config to place a SIP call to Server B... but it doesn't seem to work.

Server A (192.168.1.33) has:

exten => *136,1,Dial(SIP/90@10.10.111.13,30)

but whenever a user on Server A dials '*136' the call doesn't complete
and the CLI shows:

Executing [*136@from-sip:1] Dial("SIP/112-0071f650", "SIP/90@10.10.111.13|30") in new stack
-- Called 90@10.10.111.13
-- SIP/10.10.111.13-00793520 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

I can't see anything in Server B's logs from 192.168.1.33

What am I missing?

Any pointers to help me get this working?

--
Regards,
Russell
--------------------------------------------------------------------
| Russell Brown | MAIL: russell@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com

|
| Peterborough, England | WWW Play: http://www.ruffle.me.uk

|
--------------------------------------------------------------------

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Fw: Remove a TDM Card

Sasa wrote:
> "Tzafrir Cohen" wrote:
>
>> New:
>> loadzone=it
>> defaultzone=it
>> span=1,1,3,ccs,ami
>> bchan=1,2
>> dchan=3
>> span=2,1,3,ccs,ami
>> bchan=4-6
>> dchan=6
>>
>>> ..in zapata.conf I have:
>> ; new part:
>> switchtype=euroisdn
>> signalling = bri_net
>> priindication=outofband
>> group = 1
>> channel => 1-2
>> group = 2
>> channel => 4-5
>
> ..therefore I must only modify zaptel.conf and zapata.conf ?..and I don't
> must unload modules ?
> But when PC started without TDM card isn't a problem that is loaded
> wctdm24xxp module (that is present in rc.modules and rc.modules-2.4.33.3) on
> boot ?
> Thanks.
>

I think there's been a breakdown in terminology. You do not need to
unload the modules (rmmod wctdm24xxp). However, it sounds like you are
using Slackware, you should (but it won't hurt anything if you don't)
remove the modprobe wctdm24xxp line from your rc.modules file. If you do
not remove it the modprobe will fail because the card cannot be found
but the only result is maybe an error message on boot up.

-Dave

_______________________________________________
--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

Posted by at No comments:

Re: [asterisk-users] Do While loop

You can try something like this:

exten => _X.,1,SET(condition=${RAND(1,2)})
exten => _X.,2,GotoIf($[${condition} = '1']?1:3)
exten => _X.,3,SET(Result is 2)

Regards,
Ricardo Carvalho.
Posted by at No comments:

Re: [asterisk-users] Suppressing certain queue announcement voice prompts

> asterisk-users@rogg.is wrote:
> >> asterisk-users@rogg.is wrote:
> >>> Short of replacing a sound file with a sound file containing only a
> >>> short period of silence, is there any way to suppress certain
> sounds
> >>> from playing during queue processing by configuring for example
> >>> queues.conf or other similar files?
> >> Which announcements are you trying to not play?
> >
> > queue-thankyou for instance, to name one. Or any other of the queue-*
> > files in general. From time to time it can be convenient to change
> the
> > exact prompts played (order and contents) due to language differences
> > and personal preference of the end-users.
> >
> > We're doing this now by replacing them with silence but I'm just
> > thinking that it would be more elegant to have Asterisk not attempt
> to
> > play them in the first place. We've also removed the files in some
> > instances but that's even worse from my point of view because then we
> > get file-not-present warnings.
>
> The sounds used are configurable in queues.conf. For instance, if you
> wanted to change queue-thankyou to play something else, you could add
> the line
>
> queue-thankyou = mythankyoufile
>
> inside a queue context. Unfortunately, the order the files are played
> in is not configurable. If you don't want sounds played at all, then
> there are certain options which you can simply not set inside a queue
> in order to not have the sounds play. If you don't set a periodic-
> announce-frequency, then periodic announcements will not play.
> Similarly, if you do not set an announce-frequency, then
> position/holdtime announcements will not be played.

Well described and I understand that perfectly. The orignal point however
was if it is possible to tell the queue application to not bother with
certain announcements. I was hunting for some configuration options that are
either not present in the queues.conf sample file or perhaps that I could
find this in some totally different file that I may not have thought of
already. Not because it's unclear how to replace them (as you described very
well) with for instance a file containing very short silence or configure
the queue so that they are not applicable (like the periodic announcement),
but just to not spend time and resources on playing a file that we would
rather not hear.

Thank you for your clear reply though, you make an excellent point regarding
the existing configuration options.

_______________________________________________
--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

Posted by at No comments:
Subscribe to: Posts (Atom)