[asterisk-users] Quintum tenor configuration with asterisk help
c.savinovich at itntelecom.com
c.savinovich at itntelecom.com
Sun Sep 10 07:41:17 MST 2006
I don't have much details on your set-up, but I assume that since
quintums had performance troubles with SIP (about 2 years ago) your best
bet is to get them to work with h323. For that your first step willl be
to install h323 support on your asterisk box. I may be a little rusty
on this, so if anyone has better advice, welcome
CS
> Hi I need help configuring a quintum box with asterisk. Anyone has it
> working ?
> Thanks,
> Please let me know what I should do.
> I want to be able to register the asm200 with an extension, and be able to
> hopoff calls when calling from my asterisk,
> Thanks,
>
>
>
> On 9/9/06 6:47 PM, "asterisk-users-request at lists.digium.com"
> <asterisk-users-request at lists.digium.com> wrote:
>
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>> Today's Topics:
>>
>> 1. Re: Another (quick) Polycom 501 question (Kevin Smith)
>> 2. RE: asterisk-users Digest, Vol 26, Issue 54
>> (FRANCISCO PEREZ-LANDAETA)
>> 3. Re: Call Processing Slow 11 seconds (broadbandvoice at comcast.net)
>> 4. Re: Zaptel-1.2.9 compile error (Samy Antoun)
>> 5. Problems configuring Polycom 301 (Jim Freeze)
>> 6. Re: Zaptel-1.2.9 compile error (Nigel Godfrey)
>> 7. ztdummy installed but choppy audio warning on load (Nigel Godfrey)
>> 8. Re: ztdummy installed but choppy audio warning on load
>> (Daniel Pocock)
>> 9. Re: Zaptel-1.2.9 compile error (Samy Antoun)
>> 10. Scope of contexts (Rene)
>> 11. Re: What don't I get about SIP? (John Marvin)
>> 12. Re: Scope of contexts (Doug Lytle)
>> 13. Re: Scope of contexts (Moises Silva)
>> 14. Re: Grandstream GX-2000, doesn't send calls to free lines
>> (Zeeshan Zakaria)
>> 15. Re: How to send correct Caller ID on PRI (Zeeshan Zakaria)
>> 16. Re: How to use Grandstream GX-2000 phones for paging
>> (Zeeshan Zakaria)
>> 17. Re: Grandstream, how to use the configuration tool
>> (Zeeshan Zakaria)
>> 18. Re: Roundrobin not working on PRI (Zeeshan Zakaria)
>> 19. Using option 'r' in queue doesn't announce frequeny etc.
>> (Zeeshan Zakaria)
>>
>>
>> ----------------------------------------------------------------------
>>
>> Message: 1
>> Date: Sat, 09 Sep 2006 15:24:44 -0400
>> From: Kevin Smith <kevin.smith at mercury.net>
>> Subject: Re: [asterisk-users] Another (quick) Polycom 501 question
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <450314FC.6020309 at mercury.net>
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>> Hi Mike,
>>
>> As far as I know, you need to at least start the dialing (ie New call,
>> speaker, etc) for the digitmap to even come into play.
>>
>> The only settings that I am aware of that you can try to change are
>> dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf.
>>
>> Kevin
>>
>> Mike wrote:
>>> Hi all,
>>>
>>> That's my last one for a while (I hope).
>>>
>>> How can I (if at all possible) make the 501 turn on the speaker phone
>>> as soon as a digit is dialed (if the handset is not lifted)? Sort of
>>> like what a normal speakerphone does.
>>>
>>> The reason I want this is I want the 501 digitmap to be taken into
>>> consideration even if the handset isnt lifted and the speakerphone
>>> button isn't consciously pressed. For all those users who don't want
>>> to press send, but like dialing without lifting the handset (and can't
>>> be bothered to press the speakerphone button). Yes I know it's
>>> capricious, but we have the users we have...
>>>
>>> Yes, I have read the admin manual, but couldn't find the info. I am
>>> assuming I just don't know what to look for, but that this
>>> functionality exists.
>>>
>>>
>>>
>>> Mike
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> --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: 2
>> Date: Sat, 09 Sep 2006 19:48:27 +0000
>> From: "FRANCISCO PEREZ-LANDAETA" <fplandae at hotmail.com>
>> Subject: [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54
>> To: asterisk-users at lists.digium.com
>> Message-ID: <BAY121-F6F4A2CCA0738FB64A9537DA340 at phx.gbl>
>> Content-Type: text/plain; format=flowed
>>
>>
>> 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
>>
>>> From: asterisk-users-request at lists.digium.com
>>> Reply-To: asterisk-users at lists.digium.com
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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"
>>>
>>> Skipped content of type multipart/alternative-------------- next part
>>> --------------
>>> An embedded message was scrubbed...
>>> 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
>>> Size: 2109
>>> Url:
>>> http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/828bebd
>>> d/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
>>> -------------- next part --------------
>>> An HTML attachment was scrubbed...
>>> URL:
>>> http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/632afcb
>>> 4/attachment-0001.htm
>>>
>>> ------------------------------
>>>
>>> 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.n
>>> et>
>>>
>>> Content-Type: text/plain; charset="us-ascii"
>>>
>>> Skipped content of type multipart/alternative-------------- next part
>>> --------------
>>> An embedded message was scrubbed...
>>> 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/a805146
>>> 5/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
>>> -------------- next part --------------
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>>> 4/attachment-0001.htm
>>>
>>> ------------------------------
>>>
>>> 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?
>>>>>>
>>>>>>
>>>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
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>>>>>>>
>>>>>>> asterisk-users mailing list
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>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
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>>>>>> asterisk-users mailing list
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>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>
>>>>> asterisk-users mailing list
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>>>>>
>>>>>
>>>> _______________________________________________
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>>> -------------- next part --------------
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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
>>>> _______________________________________________
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>>>>
>>>> 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
>>>
>>>
>>> End of asterisk-users Digest, Vol 26, Issue 54
>>> **********************************************
>>
>> _________________________________________________________________
>> Check the weather nationwide with MSN Search: Try it now!
>> http://search.msn.com/results.aspx?q=weather&FORM=WLMTAG
>>
>>
>>
>> ------------------------------
>>
>> Message: 3
>> Date: Sat, 09 Sep 2006 20:27:38 +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:
>> <090920062027.15791.450323BA0005210700003DAF220682469308010B020E9B02 at comcast.n
>> et>
>>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> Thanks, I tried that and did not work for me. My users are calling US
>> number
>> and without the # at the end of the last digit dials it takes 11 seconds
>> before it starts ringing.
>>
>> -------------- Original message --------------
>> From: Alberto Sagredo <asagredo at peoplecall.com>
>>
>>> 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"
>>>> 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"
>>>> Fecha:
>>>> Sat, 9 Sep 2006 17:20:05 +0000
>>>> Para:
>>>> "Asterisk Users Mailing List - Non-Commercial Discussion"
>>>>
>>>>
>>>> Para:
>>>> "Asterisk Users Mailing List - Non-Commercial Discussion"
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> --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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>>> http://lists.digium.com/mailman/listinfo/asterisk-users
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>>
>> ------------------------------
>>
>> Message: 4
>> Date: Sat, 9 Sep 2006 13:41:43 -0700 (PDT)
>> From: Samy Antoun <samyantoun at yahoo.com>
>> Subject: Re: [asterisk-users] Zaptel-1.2.9 compile error
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <20060909204144.36193.qmail at web50115.mail.yahoo.com>
>> Content-Type: text/plain; charset=iso-8859-1
>>
>> --- Bill Maidment <bill at maidment.com.au> wrote:
>>
>>> Hi
>>> I've just tried to compile the zaptel-1.2.9 release and I get the
>>> following error:
>>
>>
>> Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors
>> when
>> compiling zap:
>>
>> make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not
>> found
>> make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not
>> found
>> make[3]: *** No rule to make target
>> `/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h',
>> needed
>> by `/usr/src/zaptel/wct4xxp/vpm450m.o'. Stop.
>> make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2
>> make[1]: *** [_module_/usr/src/zaptel] Error 2
>> make: *** [linux26] Error 2
>>
>> Hope someone has a workaround for this problem
>>
>>
>> __________________________________________________
>> Do You Yahoo!?
>> Tired of spam? Yahoo! Mail has the best spam protection around
>> http://mail.yahoo.com
>>
>>
>> ------------------------------
>>
>> Message: 5
>> Date: Sat, 9 Sep 2006 15:46:07 -0500
>> From: "Jim Freeze" <asterisk at freeze.org>
>> Subject: [asterisk-users] Problems configuring Polycom 301
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> Message-ID:
>> <5cd596d60609091346w21acea06ic1107c99cff8e30e at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi
>>
>> I have successfully been running with several Polycom SoundPoint 501
>> phones and recently purchased some Polycom 301 phones.
>> However, I can't seem to get the phones to register. The phone sees
>> the asterisk server, but all calls our are busy.
>>
>> The only difference for 'sip show peer xxx' for a working 501 phone and
>> a non working 301 phone is:
>> asterisk1*CLI>
>>
>> Addr->IP : 192.168.80.204 Port 5060 # 501
>>
>> Addr->IP : (Unspecified) Port 0 # 301
>>
>> 'sip show peers' returns:
>>
>> asterisk1*CLI> sip show peers
>> Name/username Host Dyn Nat ACL Port Status
>> 720/720 (Unspecified) D 0 UNKNOWN
>> 712/712 192.168.8.205 D 5060 OK (80
>> ms)
>> 711/711 192.168.8.203 D 5060 OK (84
>> ms)
>> 710/710 192.168.8.204 D 5060 OK (98
>> ms)
>>
>> Any 301 configuration tips would be appreciated.
>>
>> Thanks
>
>
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