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Today's Topics:
1. Re: Call Forwarding in SIP.conf ([EMAIL PROTECTED])
2. RE: Call Processing Slow 11 seconds (G.Jacobsen)
3. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock)
4. RE: Call Processing Slow 11 seconds ([EMAIL PROTECTED])
5. Re: Call Processing Slow 11 seconds (Alberto Sagredo)
6. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee)
7. Re: What don't I get about SIP? (John Marvin)
8. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock)
9. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee)
10. RE: What don't I get about SIP? (Mike)
----------------------------------------------------------------------
Message: 1
Date: Sat, 09 Sep 2006 17:12:54 +0000
From: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Call Forwarding in SIP.conf
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
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Subject: Re: [asterisk-users] Call Forwarding in SIP.conf
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Message: 2
Date: Sat, 9 Sep 2006 19:17:23 +0300
From: "G.Jacobsen" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
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In case you use an adapter or voip phone: Did you try to press hash # after
the number ? - then the adapter/voip phone dials immediately and doesnt
wait
for the next digit timeout.
Cheers
Gerry
-----Original Message----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Samstag, 9. September 2006 15:15
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Processing Slow 11 seconds
I'm having some slowness issue with Asterisk. When a number is dialed it
takes 11 seconds before it rings out. I been considering using openser for
the call processing and leaving asterisk for voicemail and conference
bridge. I get a dialtone rightaway when the receiver is picked up but after
dialing the number but within asterisk extensions and pstn numbers takes 11
seconds before ringing out. Anyone else experiencing this. I use Asterisk
1.2.3
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Message: 3
Date: Sat, 09 Sep 2006 18:23:37 +0100
From: Daniel Pocock <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
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Jason Lee wrote:
> Hi,
>
> I was testing the intel based G729 codec on SVN-trunk-r42453 following
> the
> new instructions for compiling with SVN trunk and it in preliminary
> tests it
> works ok for some calls but I found when one end of the call is an IVR
or
> Music On Hold the sound gets all distorted and asterisk segfaults. You
> can
> view the backtrace at http://pastebin.ca/165220
>
> Any assistance on this would be appreciated.
>
Have you compiled with debugging symbols instead of CPU optimization?
Can you type `bt' after the segfault, to give us some more detail?
How long into the call does this happen?
>------------------------------------------------------------------------
>
>_______________________________________________
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------------------------------
Message: 4
Date: Sat, 09 Sep 2006 17:27:15 +0000
From: [EMAIL PROTECTED]
Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
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From: "G.Jacobsen" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
Date: Sat, 9 Sep 2006 17:20:05 +0000
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------------------------------
Message: 5
Date: Sat, 09 Sep 2006 19:47:23 +0200
From: Alberto Sagredo <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Call Processing Slow 11 seconds
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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Yes you could script a dialplan putting xxxx... and S0 (zero) at the end.
An example :
(xxxxxxS0) It will dial 6 digits directly when you enter the 6th.
You could learn how to adapt your Linksys dialplan looking this wiki.
http://voip.wikispaces.com/
[EMAIL PROTECTED] escribió:
> Yes that works. I'm using Linksys adapter, is there a code I can put
> in the dial plan to prevent users from putting # after the number? I
> have a lot of people on the server and cannot ask them all to be
> pushing # after every call. Thanks for the tip and any help will be
> appreciated.
>
>
> -------------- Original message --------------
> From: "G.Jacobsen" <[EMAIL PROTECTED]>
> In case you use an adapter or voip phone: Did you try to press
> hash # after the number ? - then the adapter/voip phone dials
> immediately and doesnt wait for the next digit timeout.
>
> Cheers
>
> Gerry
>
>
> -----Original Message----
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of
> [EMAIL PROTECTED]
> *Sent:* Samstag, 9. September 2006 15:15
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Call Processing Slow 11 seconds
>
> I'm having some slowness issue with Asterisk. When a number is
> dialed it takes 11 seconds before it rings out. I been
> considering using openser for the call processing and leaving
> asterisk for voicemail and conference bridge. I get a dialtone
> rightaway when the receiver is picked up but after dialing the
> number but within asterisk extensions and pstn numbers takes
> 11 seconds before ringing out. Anyone else experiencing this.
> I use Asterisk 1.2.3
>
>
> ------------------------------------------------------------------------
>
> Asunto:
> RE: [asterisk-users] Call Processing Slow 11 seconds
> De:
> "G.Jacobsen" <[EMAIL PROTECTED]>
> Fecha:
> Sat, 9 Sep 2006 17:20:05 +0000
> Para:
> "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
>
> Para:
> "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 6
Date: Sat, 9 Sep 2006 13:03:32 -0500
From: "Jason Lee" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
I recompiled with debuging options...
both bt and btfull outputs http://pastebin.ca/165250
Before I recompiled it gave me a second of audio then I got nothing but
distortion for 5 seconds then asterisk would crash.
I retested after compiling it with just a call between two local devices
one
using ulaw and the other using g729 and I'm getting nothing but distortion.
I then tried calling music on hold and it took 3 minutes to crash the whole
time I got nothing but distortion.
On 9/9/06, Daniel Pocock <[EMAIL PROTECTED]> wrote:
>
>
>
> Jason Lee wrote:
>
> > Hi,
> >
> > I was testing the intel based G729 codec on SVN-trunk-r42453 following
> > the
> > new instructions for compiling with SVN trunk and it in preliminary
> > tests it
> > works ok for some calls but I found when one end of the call is an IVR
> or
> > Music On Hold the sound gets all distorted and asterisk segfaults. You
> > can
> > view the backtrace at http://pastebin.ca/165220
> >
> > Any assistance on this would be appreciated.
> >
> Have you compiled with debugging symbols instead of CPU optimization?
>
> Can you type `bt' after the segfault, to give us some more detail?
>
> How long into the call does this happen?
>
>
>
>------------------------------------------------------------------------
> >
> >_______________________________________________
> >--Bandwidth and Colocation provided by Easynews.com --
> >
> >asterisk-users mailing list
> >To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> _______________________________________________
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--
Regards,
Jason Lee
OmegaServ
[EMAIL PROTECTED]
Direct Line: (204) 480-1238
Toll Free: (866) 664-7786 Ext 200
http://www.omegaserv.com
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Message: 7
Date: Sat, 09 Sep 2006 12:04:33 -0600
From: John Marvin <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] What don't I get about SIP?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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Mike wrote:
> Did I misread the Asterisk wiki pages, because I believed that when a
> pattern was present, the pattern takes precedence over any "real"
> extensions? (i.e. if I have both 1234 and _1XXX as extensions in a
context)?
It's the opposite. Asterisk always uses the most specific match for an
extension, i.e. anything that matches _1XXX will take precedence over
_XXXX, but if it matches _12XX that will take precedence over _1XXX, etc.
John
------------------------------
Message: 8
Date: Sat, 09 Sep 2006 19:15:31 +0100
From: Daniel Pocock <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Jason Lee wrote:
> I recompiled with debuging options...
>
> both bt and btfull outputs http://pastebin.ca/165250
> Before I recompiled it gave me a second of audio then I got nothing but
> distortion for 5 seconds then asterisk would crash.
> I retested after compiling it with just a call between two local
> devices one
> using ulaw and the other using g729 and I'm getting nothing but
> distortion.
> I then tried calling music on hold and it took 3 minutes to crash the
> whole
> time I got nothing but distortion.
>
This suggests that someone/something gave the command `stop now'
Can you send the backtrace from a segfault?
>
> On 9/9/06, Daniel Pocock <[EMAIL PROTECTED]> wrote:
>
>>
>>
>>
>> Jason Lee wrote:
>>
>> > Hi,
>> >
>> > I was testing the intel based G729 codec on SVN-trunk-r42453
following
>> > the
>> > new instructions for compiling with SVN trunk and it in preliminary
>> > tests it
>> > works ok for some calls but I found when one end of the call is an
IVR
>> or
>> > Music On Hold the sound gets all distorted and asterisk segfaults.
You
>> > can
>> > view the backtrace at http://pastebin.ca/165220
>> >
>> > Any assistance on this would be appreciated.
>> >
>> Have you compiled with debugging symbols instead of CPU optimization?
>>
>> Can you type `bt' after the segfault, to give us some more detail?
>>
>> How long into the call does this happen?
>>
>>
>>
>------------------------------------------------------------------------
>>
>> >
>> >_______________________________________________
>> >--Bandwidth and Colocation provided by Easynews.com --
>> >
>> >asterisk-users mailing list
>> >To UNSUBSCRIBE or update options visit:
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.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
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>
------------------------------
Message: 9
Date: Sat, 9 Sep 2006 13:28:55 -0500
From: "Jason Lee" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
Sorry about that. I thought I had the right core dump. I retried again and
the output from bt and bt full is at http://pastebin.ca/165289
It took 1min 50seconds of nothing but distortion before asterisk segfaulted
--
Regards,
Jason
On 9/9/06, Daniel Pocock <[EMAIL PROTECTED]> wrote:
>
>
>
> Jason Lee wrote:
>
> > I recompiled with debuging options...
> >
> > both bt and btfull outputs http://pastebin.ca/165250
> > Before I recompiled it gave me a second of audio then I got nothing
but
> > distortion for 5 seconds then asterisk would crash.
> > I retested after compiling it with just a call between two local
> > devices one
> > using ulaw and the other using g729 and I'm getting nothing but
> > distortion.
> > I then tried calling music on hold and it took 3 minutes to crash the
> > whole
> > time I got nothing but distortion.
> >
> This suggests that someone/something gave the command `stop now'
>
> Can you send the backtrace from a segfault?
>
> >
> > On 9/9/06, Daniel Pocock <[EMAIL PROTECTED]> wrote:
> >
> >>
> >>
> >>
> >> Jason Lee wrote:
> >>
> >> > Hi,
> >> >
> >> > I was testing the intel based G729 codec on SVN-trunk-r42453
> following
> >> > the
> >> > new instructions for compiling with SVN trunk and it in preliminary
> >> > tests it
> >> > works ok for some calls but I found when one end of the call is an
> IVR
> >> or
> >> > Music On Hold the sound gets all distorted and asterisk segfaults.
> You
> >> > can
> >> > view the backtrace at http://pastebin.ca/165220
> >> >
> >> > Any assistance on this would be appreciated.
> >> >
> >> Have you compiled with debugging symbols instead of CPU optimization?
> >>
> >> Can you type `bt' after the segfault, to give us some more detail?
> >>
> >> How long into the call does this happen?
> >>
> >>
> >>
>
>------------------------------------------------------------------------
> >>
> >> >
> >> >_______________________________________________
> >> >--Bandwidth and Colocation provided by Easynews.com --
> >> >
> >> >asterisk-users mailing list
> >> >To UNSUBSCRIBE or update options visit:
> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >
> >> >
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.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
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> >
> >
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Message: 10
Date: Sat, 9 Sep 2006 14:58:32 -0400
From: "Mike" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] What don't I get about SIP?
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
It certainly makes sense, and I tried it...it works, you are right.
So what do you make of this page :
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting
Mike
> -----Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> John Marvin
> Sent: September 9, 2006 2:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] What don't I get about SIP?
>
> Mike wrote:
>
> > Did I misread the Asterisk wiki pages, because I believed
> that when a
> > pattern was present, the pattern takes precedence over any "real"
> > extensions? (i.e. if I have both 1234 and _1XXX as
> extensions in a context)?
>
> It's the opposite. Asterisk always uses the most specific
> match for an extension, i.e. anything that matches _1XXX will
> take precedence over _XXXX, but if it matches _12XX that will
> take precedence over _1XXX, etc.
>
> John
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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