Re: [asterisk-users] g729 on 1.4.10.1
On 9/20/07, Luke Groeneveld <[EMAIL PROTECTED]> wrote: > > > I'm getting frustrated simply trying to get this g729 working. > > For what it is worth, I had a similar issue to you, and managed to get > g729 working by installing the binary files from http://asterisk.hosting.lv > Thanks for the suggestion. Looks like I'm having the same problem, though. What's odd is that I can make phone to phone G729 calls through Asterisk, but G729 calls from my gateway do not work. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
I'm trying a simple Echo test and it's failing for g729... exten => 1267,1,Answer() exten => 1267,2,Echo() Test #1 (failure) gateway33 codecs g729a, g729b [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=g729 gateway33 INVITE = g729b Asterisk 200 OK = no media Asterisk sends (one way) g711u RTP Test #2 (failure) gateway33 codecs g729a [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=g729 gateway33 INVITE = g729a Asterisk 200 OK = no media gateway33 ends the session gateway33 INVITE = g729a Asterisk 200 OK = no media gateway33 ends the session ... Test #3 (success) gateway33 codec g729a, g729b, g711u [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=ulaw allow=g729 gateway33 INVITE = g729b, g711u Asterisk 200 OK = g711u Asterisk sends/receives g711u RTP Does any of this point to a specific problem? I even have a licensed g729 channel. CLI> show g729 0/0 encoders/decoders of 1 licensed channels are currently in use What information can I provide to help troubleshoot? This is making no sense. When I setup my desk phone to use G729 and make the test call directly (bypassing the gateway), the call completes fine and media is sent using G729 successfully. I'm not sure why it would work any differently from a Cisco gateway? The only difference that I'm aware of is that my phone (Polycom 430) seemed to ask for G729, while the gateway was either G729a or G729b specifically. In the instance of my phone, Asterisk came back with G729a in the 200 OK message. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
On Thu, 20 Sep 2007 02:15:11 am Scott Moseman wrote: > I'm getting frustrated simply trying to get this g729 working. For what it is worth, I had a similar issue to you, and managed to get g729 working by installing the binary files from http://asterisk.hosting.lv ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
On 9/18/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > > > However, in Test #3 the call will fail. Why? > > Because Asterisk will attempt to use ulaw in preference to G.729 if > possible, and the other endpoint offered to support ulaw. The format(s) > supported by the eventual call destination are not relevant, because at > the time Asterisk is making a format decision for the incoming call leg, > it has no clue what the destination is going to be or what formats it > will support. > We purchased a 1 channel G729 license for testing purposes. However, I'm still having problems and G729 calls do not work. [src_gateway] disallow=all allow=g729 [dest_phone] disallow=all allow=ulaw allow=alaw The INVITE from the gatway to Asterisk contains g729b. The INVITE from Asterisk to the phone has g711u|g711a. The dest phone rings, but when picked up, the call is gone and the source phone gets a "busy" response. What's up? CLI> show g729 0/0 encoders/decoders of 1 licensed channels are currently in use I'm getting frustrated simply trying to get this g729 working. Where do I need to increase debugging in order to find out more detailed and useful info about what's going wrong??? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Scott Moseman wrote: > However, in Test #3 the call will fail. Why? Because Asterisk will attempt to use ulaw in preference to G.729 if possible, and the other endpoint offered to support ulaw. The format(s) supported by the eventual call destination are not relevant, because at the time Asterisk is making a format decision for the incoming call leg, it has no clue what the destination is going to be or what formats it will support. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Follow me on this, it seems odd (or maybe I don't undertand)... Test #1 [src_phone] disallow=all allow=g729 [dest_phone] disallow=all allow=g729 I can make the call (src to dest) and it will work using g729. Both the call handling and media are going through Asterisk. Test #2 [src_phone] disallow=all allow=g729 allow=ulaw [dest_phone] disallow=all allow=g729 I can make the call (src to dest) and it will work using g729. Both the call handling and media are going through Asterisk. Test #3 [src_phone] disallow=all allow=ulaw allow=g729 [dest_phone] disallow=all allow=g729 The above call attempt will fail, and this is what I'm seeing: chan_sip.c:2944 sip_call: No audio format found to offer. In every test, the source INVITE includes ulaw, alaw and 729. That is the codecs that I configured on the phone themselves. However, in Test #3 the call will fail. Why? This does not necessarily have to do with my g729 gateway, but I'm curious what's wrong with this scenario, maybe using this situation to understand will help me with my gateway... (Although I tried setting only g729 on the gateway and the gateway's peer in the Asterisk and it did not appear to help.) Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Scott Moseman wrote: > The gateway is transcoding the PSTN into g729 and passing it to > Asterisk. The Asterisk never sees the PSTN from the outside. I have > watched the INVITE requests, they contain a request for a g729 only > call. But the INVITE to the phone does not include g729. > > However, as previously stated, I did get a g729 phone to talk to > another g729 phone. So I assume that means pass-through *can* work, > but something is not working right? > > Thanks, > Scott > If you have anything in Asterisk trying to handle the audio, you cannot pass it through. For instance, if you are trying to record the call in ulaw, or trying to playback prompts that aren't available in g729. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Matt Watson wrote: > > -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman >> >> My understanding was that it's not required for pass-through. >> >> PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone >> >> Does this not equate to pass-through? Maybe I misunderstood? > PSTN -> g729 requires transcoding at that point. > > You can however do: > > G.729 phone -> asterisk -> G.729 phone without license (from my > understanding). > > But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it > requires a license to preform transcoding. Matt, Look at his path. He's going from a PSTN phone to a g729 gateway. As long as the gateway is there, Asterisk doesn't really know about the PSTN phone. Therefore, yes, this should equate to pass through. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
The gateway is transcoding the PSTN into g729 and passing it to Asterisk. The Asterisk never sees the PSTN from the outside. I have watched the INVITE requests, they contain a request for a g729 only call. But the INVITE to the phone does not include g729. However, as previously stated, I did get a g729 phone to talk to another g729 phone. So I assume that means pass-through *can* work, but something is not working right? Thanks, Scott On 9/18/07, Matt Watson <[EMAIL PROTECTED]> wrote: > > PSTN -> g729 requires transcoding at that point. > > You can however do: > > G.729 phone -> asterisk -> G.729 phone without license (from my > understanding). > > But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) > it requires a license to preform transcoding. > > -- > Matt > > -Original Message- > From: [EMAIL PROTECTED] > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] g729 on 1.4.10.1 > > On 9/18/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > > > > I hate to ask what may be a silly question, but have you purchased > > any G.729 licenses to use with the g.729 codec you downloaded? > > If you haven't registered codec_g729 yet, that would be why you are > > seeing this problem with codec_g729. > > > > My understanding was that it's not required for pass-through. > > PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone > > Does this not equate to pass-through? Maybe I misunderstood? > > Thanks, > Scott > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Fyi... [myphone] disallow=all allow=g729 canreinvite=no [otherphone] disallow=all allow=g729 canreinvite=no I attempted this setup and it works. Media routed through the Asterisk. Thanks, Scott On 9/18/07, Jeremy Mann <[EMAIL PROTECTED]> wrote: > > Does G.729 phone -> asterisk -> G.729 phone work with reinvite turned off? > > -Original Message- > From: [EMAIL PROTECTED] > Sent: Tuesday, September 18, 2007 1:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] g729 on 1.4.10.1 > > PSTN -> g729 requires transcoding at that point. > > You can however do: > > G.729 phone -> asterisk -> G.729 phone without license (from my > understanding). > > But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it > requires a license to preform transcoding. > > -- > Matt > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman > Sent: September-18-07 1:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] g729 on 1.4.10.1 > > On 9/18/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > > > > I hate to ask what may be a silly question, but have you purchased > > any G.729 licenses to use with the g.729 codec you downloaded? > > If you haven't registered codec_g729 yet, that would be why you are > > seeing this problem with codec_g729. > > > > My understanding was that it's not required for pass-through. > > PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone > > Does this not equate to pass-through? Maybe I misunderstood? > > Thanks, > Scott > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Does G.729 phone -> asterisk -> G.729 phone work with reinvite turned off? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson Sent: Tuesday, September 18, 2007 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 PSTN -> g729 requires transcoding at that point. You can however do: G.729 phone -> asterisk -> G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > > I hate to ask what may be a silly question, but have you purchased > any G.729 licenses to use with the g.729 codec you downloaded? > If you haven't registered codec_g729 yet, that would be why you are > seeing this problem with codec_g729. > My understanding was that it's not required for pass-through. PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
PSTN -> g729 requires transcoding at that point. You can however do: G.729 phone -> asterisk -> G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > > I hate to ask what may be a silly question, but have you purchased > any G.729 licenses to use with the g.729 codec you downloaded? > If you haven't registered codec_g729 yet, that would be why you are > seeing this problem with codec_g729. > My understanding was that it's not required for pass-through. PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
I am experiencing the exact same problem on solaris, and we do have licenses purchased. I will log a bug at digium in the next day or two about my particular instance. Scott Moseman wrote: > On 9/18/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > >> I hate to ask what may be a silly question, but have you purchased >> any G.729 licenses to use with the g.729 codec you downloaded? >> If you haven't registered codec_g729 yet, that would be why you are >> seeing this problem with codec_g729. >> >> > > My understanding was that it's not required for pass-through. > > PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone > > Does this not equate to pass-through? Maybe I misunderstood? > > Thanks, > Scott > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > begin:vcard fn:Bruce McAlister n:McAlister;Bruce org:Blueface Ltd adr:;;8 Clanwilliam Terrace;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 524 2009 x-mozilla-html:FALSE url:http://www.blueface.ie version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
On 9/18/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > > I hate to ask what may be a silly question, but have you purchased > any G.729 licenses to use with the g.729 codec you downloaded? > If you haven't registered codec_g729 yet, that would be why you are > seeing this problem with codec_g729. > My understanding was that it's not required for pass-through. PSTN Phone -> g729 Gateway -> Asterisk -> g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Scott Moseman wrote: > Here's what I'm showing in the logs... > > [Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format > g729, extension(s) g729 > [Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so => (Raw G729 data) > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module > version 32, Copyright (C) 1999-2007 Digium, Inc. > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This module is supplied > under a commercial license granted by Digium, Inc. > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Please see the full > license text supplied by the accompanying > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: "register" utility, or > ask for a copy from Digium. > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This product includes > software developed by the OpenSSL Project > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: for use in the OpenSSL > Toolkit. (http://www.openssl.org/) > [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Copyright (C) 1998-2006 > The OpenSSL Project > [Sep 18 09:52:09] VERBOSE[2786] logger.c: == G.729 Host-ID: x:x:x:etc > [Sep 18 09:52:09] WARNING[2786] codec_g729.c: Failed to initialize > G.729 copy protection! > [Sep 18 09:52:09] VERBOSE[2786] logger.c: codec_g729a.so => (Annex A/B > (floating point) G.729 Codec (optimized for i686)) > > Any ideas where this points me? I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. Matthew Fredrickson > > Thanks, > Scott > > > > On 9/17/07, Scott Moseman <[EMAIL PROTECTED]> wrote: >> What's the best way to debug what's going on within Asterisk? >> I turned up the 'core debug', but that did not give me what I was >> hoping to find. I'm hoping to see some kind of error that explains >> why it will not pass through the g729 codec. >> >> Thanks, >> Scott >> >> >> On 9/14/07, Scott Moseman <[EMAIL PROTECTED]> wrote: >>> I have a fresh 1.4.10.1 installation that appears to have a problem >>> with g729 pass-through. I can see the gateway in question sending >>> an INVITE using g729b. However, the Asterisk is only sending g711 >>> in the INVITE to my Polycom phone. >>> >>> [gateway] >>> disallow=all >>> allow=g729 >>> >>> [phone] >>> disallow=all >>> allow=ulaw >>> allow=alaw >>> allow=g729 >>> >>> There's the codec configs for the gateway and the phone in question. >>> I even attempted to setup the phone with only the allow=g729, but in >>> that instance it won't even complete the call. We had to add g711 >>> support to the gateway in question for now to get it up and running, >>> but we want to get it back to using only g729. >>> >>> CLI> show modules like g729 >>> Module Description >>> Use Count >>> format_g729.so Raw G729 data >>> 0 >>> codec_g729a.so Annex A/B (floating point) G.729 Codec >>> ( 0 >>> 2 modules loaded >>> >>> I downloaded the pre-compiled g729 module from Digium. The sip.conf >>> references g729 and the codec module is loaded. Unless there's >>> anything else I need to do that I'm missing? >>> >>> Thanks, >>> Scott >>> > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Here's what I'm showing in the logs... [Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format g729, extension(s) g729 [Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so => (Raw G729 data) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This module is supplied under a commercial license granted by Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Please see the full license text supplied by the accompanying [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: "register" utility, or ask for a copy from Digium. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This product includes software developed by the OpenSSL Project [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Copyright (C) 1998-2006 The OpenSSL Project [Sep 18 09:52:09] VERBOSE[2786] logger.c: == G.729 Host-ID: x:x:x:etc [Sep 18 09:52:09] WARNING[2786] codec_g729.c: Failed to initialize G.729 copy protection! [Sep 18 09:52:09] VERBOSE[2786] logger.c: codec_g729a.so => (Annex A/B (floating point) G.729 Codec (optimized for i686)) Any ideas where this points me? Thanks, Scott On 9/17/07, Scott Moseman <[EMAIL PROTECTED]> wrote: > > What's the best way to debug what's going on within Asterisk? > I turned up the 'core debug', but that did not give me what I was > hoping to find. I'm hoping to see some kind of error that explains > why it will not pass through the g729 codec. > > Thanks, > Scott > > > On 9/14/07, Scott Moseman <[EMAIL PROTECTED]> wrote: > > > > I have a fresh 1.4.10.1 installation that appears to have a problem > > with g729 pass-through. I can see the gateway in question sending > > an INVITE using g729b. However, the Asterisk is only sending g711 > > in the INVITE to my Polycom phone. > > > > [gateway] > > disallow=all > > allow=g729 > > > > [phone] > > disallow=all > > allow=ulaw > > allow=alaw > > allow=g729 > > > > There's the codec configs for the gateway and the phone in question. > > I even attempted to setup the phone with only the allow=g729, but in > > that instance it won't even complete the call. We had to add g711 > > support to the gateway in question for now to get it up and running, > > but we want to get it back to using only g729. > > > > CLI> show modules like g729 > > Module Description > > Use Count > > format_g729.so Raw G729 data > > 0 > > codec_g729a.so Annex A/B (floating point) G.729 Codec > > ( 0 > > 2 modules loaded > > > > I downloaded the pre-compiled g729 module from Digium. The sip.conf > > references g729 and the codec module is loaded. Unless there's > > anything else I need to do that I'm missing? > > > > Thanks, > > Scott > > > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec. Thanks, Scott On 9/14/07, Scott Moseman <[EMAIL PROTECTED]> wrote: > > I have a fresh 1.4.10.1 installation that appears to have a problem > with g729 pass-through. I can see the gateway in question sending > an INVITE using g729b. However, the Asterisk is only sending g711 > in the INVITE to my Polycom phone. > > [gateway] > disallow=all > allow=g729 > > [phone] > disallow=all > allow=ulaw > allow=alaw > allow=g729 > > There's the codec configs for the gateway and the phone in question. > I even attempted to setup the phone with only the allow=g729, but in > that instance it won't even complete the call. We had to add g711 > support to the gateway in question for now to get it up and running, > but we want to get it back to using only g729. > > CLI> show modules like g729 > Module Description > Use Count > format_g729.so Raw G729 data > 0 > codec_g729a.so Annex A/B (floating point) G.729 Codec > ( 0 > 2 modules loaded > > I downloaded the pre-compiled g729 module from Digium. The sip.conf > references g729 and the codec module is loaded. Unless there's > anything else I need to do that I'm missing? > > Thanks, > Scott > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g729 on 1.4.10.1
I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI> show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
