[asterisk-users] Quintum tenor configuration with asterisk help

FRANCISCO PEREZ-LANDAETA fplandae at hotmail.com
Sat Sep 9 22:09:46 MST 2006


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
>> To: asterisk-users at lists.digium.com
>> Subject: asterisk-users Digest, Vol 26, Issue 54
>> Date: Sat,  9 Sep 2006 12:00:25 -0700 (MST)
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>> Send asterisk-users mailing list submissions to
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>> 
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>> or, via email, send a message with subject or body 'help' to
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>> When replying, please edit your Subject line so it is more specific
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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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>> http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/d4e38b7
>> 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?
>>>>> 
>>>>> 
>>>>> 
>>> 
>>> ------------------------------------------------------------------------
>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> --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
>>>> 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
>>> 
>>> 
>> 
>> 
>> 
>> ------------------------------
>> 
>> _______________________________________________
>> --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
>>> 
>> 
>> _______________________________________________
>> --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: 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
> 
> 
> __________________________________________________
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> 
> ------------------------------
> 
> 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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