[asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54
FRANCISCO PEREZ-LANDAETA
fplandae at hotmail.com
Sat Sep 9 12:48:27 MST 2006
hi i need helpl configuring a quintum tenor analog gateway using sip with
asterisk.
anyone,
help is appreciated
the model of the gteway is asm200 i need the settings to configure it with
asterisk.
for some reason it registers with asterisk but when try to call the
extension from the quintum it is not recognized.
help help help
thanks
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>Today's Topics:
>
> 1. Re: Call Forwarding in SIP.conf (broadbandvoice at comcast.net)
> 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 (broadbandvoice at comcast.net)
> 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: broadbandvoice at comcast.net
>Subject: Re: [asterisk-users] Call Forwarding in SIP.conf
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID:
> <090920061712.20356.4502F61600032D6F00004F84220588644208010B020E9B02 at comcast.net>
>
>Content-Type: text/plain; charset="us-ascii"
>
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>From: "Tim St. Pierre" <tim at communicatefreely.net>
>Subject: Re: [asterisk-users] Call Forwarding in SIP.conf
>Date: Sat, 9 Sep 2006 16:52:40 +0000
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>Url:
>http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/828bebdd/attachment-0001.eml
>
>------------------------------
>
>Message: 2
>Date: Sat, 9 Sep 2006 19:17:23 +0300
>From: "G.Jacobsen" <g_jacobsen at yahoo.co.uk>
>Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
>Message-ID: <CPEBJFBCDCKKIHJAODHCCEPGCLAA.g_jacobsen at yahoo.co.uk>
>Content-Type: text/plain; charset="us-ascii"
>
>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: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of
>broadbandvoice at comcast.net
> Sent: Samstag, 9. September 2006 15:15
> To: asterisk-users at 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 <daniel at readytechnology.co.uk>
>Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID: <4502F899.4010602 at readytechnology.co.uk>
>Content-Type: text/plain; charset=us-ascii; format=flowed
>
>
>
>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
> >
> >
>
>
>------------------------------
>
>Message: 4
>Date: Sat, 09 Sep 2006 17:27:15 +0000
>From: broadbandvoice at comcast.net
>Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID:
> <090920061727.5745.4502F9730006E06300001671220699973508010B020E9B02 at comcast.net>
>
>Content-Type: text/plain; charset="us-ascii"
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>From: "G.Jacobsen" <g_jacobsen at yahoo.co.uk>
>Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
>Date: Sat, 9 Sep 2006 17:20:05 +0000
>Size: 818
>Url:
>http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/a8051465/attachment-0001.eml
>
>------------------------------
>
>Message: 5
>Date: Sat, 09 Sep 2006 19:47:23 +0200
>From: Alberto Sagredo <asagredo at peoplecall.com>
>Subject: Re: [asterisk-users] Call Processing Slow 11 seconds
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID: <4502FE2B.1020200 at peoplecall.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>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/
>
>broadbandvoice at comcast.net 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" <g_jacobsen at yahoo.co.uk>
> > 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:* asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com]*On Behalf Of
> > *broadbandvoice at comcast.net
> > *Sent:* Samstag, 9. September 2006 15:15
> > *To:* asterisk-users at 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" <g_jacobsen at yahoo.co.uk>
> > Fecha:
> > Sat, 9 Sep 2006 17:20:05 +0000
> > Para:
> > "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> >
> > Para:
> > "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at 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" <jason.m.lee at gmail.com>
>Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
>Message-ID:
> <c3bac2490609091103l489be6bas75c63061e1a7cf4c at mail.gmail.com>
>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 <daniel at readytechnology.co.uk> 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
> >
>
>
>
>--
>Regards,
>
>Jason Lee
>OmegaServ
>jlee at omegaserv.com
>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 <jm-asterisk at themarvins.org>
>Subject: Re: [asterisk-users] What don't I get about SIP?
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID: <45030231.4060808 at themarvins.org>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>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 <daniel at readytechnology.co.uk>
>Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
>To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
>Message-ID: <450304C3.2060505 at readytechnology.co.uk>
>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 <daniel at readytechnology.co.uk> 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
> >>
> >
> >
> >
> >------------------------------------------------------------------------
> >
> >_______________________________________________
> >--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: 9
>Date: Sat, 9 Sep 2006 13:28:55 -0500
>From: "Jason Lee" <jason.m.lee at gmail.com>
>Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
>Message-ID:
> <c3bac2490609091128y4235e54dqace530af644cf1a3 at mail.gmail.com>
>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 <daniel at readytechnology.co.uk> 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 <daniel at readytechnology.co.uk> 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
> > >>
> > >
> > >
> > >
> >
> >------------------------------------------------------------------------
> > >
> > >_______________________________________________
> > >--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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>------------------------------
>
>Message: 10
>Date: Sat, 9 Sep 2006 14:58:32 -0400
>From: "Mike" <list at virtutel.ca>
>Subject: RE: [asterisk-users] What don't I get about SIP?
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
>Message-ID: <00bb01c6d441$f36800c0$0a01a8c0 at MIKE>
>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: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] 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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>End of asterisk-users Digest, Vol 26, Issue 54
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