[asterisk-users] Codec opus returned invalid number of samples

Hello, I am getting these warnings in the logs when using linphone client: WARNING: Codec opus returned invalid number of samples these are probably harmless warnings (the call works fine), but it just floods the logs unnecessarily any way to fix this problem, or suppress the warnings? --

Re: [asterisk-users] Codec question

turn off g726. On 6/17/20 4:34 PM, Jerry Geis wrote: Ok - updating the firmware on teh device - factory reset, re-config. Capabilities: us - (g726|slin16|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g726|slin16)/video=(nothing)/text=(nothing), combined - (g726|slin16|ulaw|alaw) Looking much better

Re: [asterisk-users] Codec question

Ok - updating the firmware on teh device - factory reset, re-config. Capabilities: us - (g726|slin16|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g726|slin16)/video=(nothing)/text=(nothing), combined - (g726|slin16|ulaw|alaw) Looking much better. Jerry On Wed, Jun 17, 2020 at 4:01 PM Jerry Geis wrote

Re: [asterisk-users] Codec question

I thought - what about the software - maybe it needs updated. After doing so I get a list: Audio codecs PCMU (8000 Hz) PCMA (8000 Hz) opus (48000 Hz) L16/16000 (16000 Hz) G.726-32 (8000 Hz) L16/8000 (8000 Hz) speex/16000 (16000 Hz) speex/8000 (8000 Hz) -- _

Re: [asterisk-users] Codec question

Docs said this: Audio Codecs: G.711, G.726, WAV, MP3. This is all it shows: Got SDP version 3801411990 and unique parts [- 3801411989 IN IP4 192.168.2.3] Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event

Re: [asterisk-users] Codec question

On Wed, Jun 17, 2020 at 11:13 AM Jerry Geis wrote: > I see this device : > Axis C8033 Audio Bridge Quick Specs: > Communications Protocol: SIP. > Ethernet Ports: 1x 10/100. > PoE: 802.3af/at Type 1 Class 2. > Additional Interfaces: > Audio: one-way/two-way, mono. > Audio Codecs: G.711, G.726, WAV

[asterisk-users] Codec question

I see this device : Axis C8033 Audio Bridge Quick Specs: Communications Protocol: SIP. Ethernet Ports: 1x 10/100. PoE: 802.3af/at Type 1 Class 2. Additional Interfaces: Audio: one-way/two-way, mono. Audio Codecs: G.711, G.726, WAV, MP3. Edge Storage: microSD, microSDHC, microSDXC. Operating Tempera

Re: [asterisk-users] codec opus on centos 6 with asterisk 16

Hello, Did you install the "opus" RPM ? Regards Jean Le 09/09/2019 à 13:08, Israel Gottlieb a écrit : Hi list does anyone know how i could use codec opus with asterisk 16 when using centos 6 the prebuilt binary from digium doesnt load Thanks, Israel -- _

[asterisk-users] codec opus on centos 6 with asterisk 16

Hi list does anyone know how i could use codec opus with asterisk 16 when using centos 6 the prebuilt binary from digium doesnt load Thanks, Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Chec

Re: [asterisk-users] codec negotiation or transcoding issue

Asterisk might be unable to transcode rtp type from downstream to upstream, or vice versa. There's a bug reported here, for asterisk 12 or above, using chan_sip. https://issues.asterisk.org/jira/browse/ASTERISK-25676 It says that you could avoid the bug by using chan_pjsip, but you still encounter

[asterisk-users] codec negotiation or transcoding issue

Hi, I'm facing strange issue while establishing inbound calls from SIP trunks. Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has select

Re: [asterisk-users] Codec Negotiation problem

Hi Matt Thanks for your response. I have tried with two GXV3175 with same result. Let me dig deep on this to find out the route cause Sam Matthew Jordan wrote: > On Thu, Jun 13, 2013 at 12:04 PM, wrote: > >> Hi there >> >> I have asterisk 10.11.1 which seems to have problem negotiating codec. >>

Re: [asterisk-users] Codec Negotiation problem

On Thu, Jun 13, 2013 at 12:04 PM, wrote: > Hi there > > I have asterisk 10.11.1 which seems to have problem negotiating codec. > > Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p > and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, > h263p. I have tri

[asterisk-users] Codec Negotiation problem

Hi there I have asterisk 10.11.1 which seems to have problem negotiating codec. Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, h263p. I have tried similar combination of codecs and SIP phone but when

[asterisk-users] Codec Mismatch

Sometimes in huge call volume am facing this type of error, [Jun 4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write: Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:04] WARNING[8285][C-79da]:

Re: [asterisk-users] Codec warnings after upgrade to 1.8

Well, I don't know if this has been resolved yet, but I struggled with this issue very seriously for about 12 straight hours: I had an Asterisk 1.4 box with a Digium FXO/FXS card and an Asterisk 1.8 box i just built with no cards. I migrated all extensions to the new box. All are Cisco phones using

[asterisk-users] codec priorities

Hello, I am about to start playing with wideband codecs in our lab, and was hoping to get some clarification on a few things. To date I've pretty much forced the use of G.711 on all legs of all calls, and life has been grand. Now we are distributing phones with G.722 and speex capability, and I

Re: [asterisk-users] codec mismatch on channel

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, February 22, 2012 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] codec mismatch

Re: [asterisk-users] codec mismatch on channel

February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting

Re: [asterisk-users] codec mismatch on channel

-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to

[asterisk-users] Codec

Anyone using the G729 codec to create a h.323 trunk between an Avaya Communication manager and Asterisk Freepbx System and works? I don't have the G729 codec installed on the Asterisk and running G711MU on avaya and getting invalid codec when calling from Avaya to Asterisk. -- _

Re: [asterisk-users] Codec warnings after upgrade to 1.8

On 12/27/11 18:23, Olivier wrote: 2011/12/27, Eric Wieling : We are running 1.8.8.0. Then the issue you're having differs from the one I had (which appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8 respectively). Upgrading to 1.8.8 DID NOT HELP I'm getting the same erro

Re: [asterisk-users] Codec warnings after upgrade to 1.8

2011/12/27, Eric Wieling : > We are running 1.8.8.0. > Then the issue you're having differs from the one I had (which appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8 respectively). -- _ -- Bandwidth and Colo

Re: [asterisk-users] Codec warnings after upgrade to 1.8

On 12/27/11 08:16, Ryan Wagoner wrote: On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling wrote: I'm getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)? WARNING[11123]: channel.c:4909 as

Re: [asterisk-users] Codec warnings after upgrade to 1.8

-users] Codec warnings after upgrade to 1.8 On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling wrote: I'm getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)? WARNING[

Re: [asterisk-users] Codec warnings after upgrade to 1.8

On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling wrote: > I'm getting various codec related warnings after upgrading to 1.8. Did I > miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)? > > WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel > DAHDI/i1/121242

Re: [asterisk-users] Codec warnings after upgrade to 1.8

Could you try with 1.8.8.0 ? I think this one includes a fix for that error. 2011/12/26, Joseph : > On 12/23/11 10:40, Eric Wieling wrote: >>I'm getting various codec related warnings after upgrading to 1.8. Did I >> miss something in the UPGRADE file? Does Asterisk no longer transcode >> 8-)? >

Re: [asterisk-users] Codec warnings after upgrade to 1.8

On 12/23/11 10:40, Eric Wieling wrote: I'm getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)? WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel DAHDI/i1/12124221200-74 setting write

[asterisk-users] Codec warnings after upgrade to 1.8

I'm getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)? WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native formats

[asterisk-users] Codec warning polluting the CLI since 1.8

Hi, I've just upgraded to 1.8.6 on one server and I've been getting a lot of codec warning, like this: WARNING[21211]: chan_sip.c:6341 sip_write: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729) I do have a Digium transcoder

Re: [asterisk-users] Codec negotiation issue (no audio format found to offer)

sage - >> > From: "Ryan McGuire" >> > To: asterisk-users@lists.digium.com >> > Sent: Wednesday, August 3, 2011 9:47:42 AM >> > Subject: Re: [asterisk-users] Codec negotiation issue (no audio format >> > found to offer) >> > From l

Re: [asterisk-users] Codec negotiation issue (no audio format found to offer)

On Thu, Aug 4, 2011 at 9:58 AM, David Vossel wrote: > - Original Message - > > From: "Ryan McGuire" > > To: asterisk-users@lists.digium.com > > Sent: Wednesday, August 3, 2011 9:47:42 AM > > Subject: Re: [asterisk-users] Codec negotiation issue

Re: [asterisk-users] Codec negotiation issue (no audio format found to offer)

- Original Message - > From: "Ryan McGuire" > To: asterisk-users@lists.digium.com > Sent: Wednesday, August 3, 2011 9:47:42 AM > Subject: Re: [asterisk-users] Codec negotiation issue (no audio format found > to offer) > From looking into this, it appears a

Re: [asterisk-users] Codec negotiation issue (no audio format found to offer)

>From looking into this, it appears as if this is due to Asterisk negotiating the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egres

[asterisk-users] Codec negotiation issue (no audio format found to offer)

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be u

Re: [asterisk-users] Codec translation from gsm to other codecs or from other codecs to gsm

asterisk-users@lists.digium.com > Subject: [asterisk-users] Codec translation from gsm to other codecs or from > other codecs to gsm > > Hi All; > > The asterisk version is 1.8.4.2 > > Why codec translation from and to gsm is not possible? I think it was > possible in prev

Re: [asterisk-users] Codec translation from gsm to other codecs or from other codecs to gsm

On 07/31/2011 07:48 AM, bilal ghayyad wrote: Hi All; The asterisk version is 1.8.4.2 Why codec translation from and to gsm is not possible? I think it was possible in previous versions. I am missing something to have this codec translation possibility? What gives you the impression that it

[asterisk-users] Codec translation from gsm to other codecs or from other codecs to gsm

Hi All; The asterisk version is 1.8.4.2 Why codec translation from and to gsm is not possible? I think it was possible in previous versions. I am missing something to have this codec translation possibility? Please advise. Regards Bilal --

Re: [asterisk-users] Codec negotiation

Hi, If you will send call without answering on asterisk and have directrtpsetup=yes in sip.conf codec negociation will always be between UAs so any matched codec will work fine. If you are answering call on asterisk then dialing it out to next UA then you need to add canreinvite=yes for both

[asterisk-users] Codec negotiation

Hi List, I am using asterisk 1.8.1. and I want to avoid transcoding when 2 SIP peers calling each other: A (g722, alaw) calls B (alaw,ulaw) via asterisk. My setup: allow=g722,alaw preferred_codec_only=no Note that when B calls A, codec alaw is used on both ends, fine, but it does not seem to

Re: [asterisk-users] Codec choice

This is at least the third post under the subject 'Codec Choice' by the same sender. Why don't you stay within your first thread? Does posting over and over again increases chances of getting a solution? If so, then maybe I should try the same, as seems like an increasing trend on this list. Zeesh

[asterisk-users] Codec choice

Hi, Group () and Group_Count () will need to be used on certain extension. What if there are lot of clients on the kit with different routings some going to dahdi and some to different sip interconnects, how can we do it on whole kit basis. Or let me know if there is any other way to use these

Re: [asterisk-users] Codec choice

On Fri, 20 Aug 2010, Sherwood McGowan wrote: > Good point my man...You drinking yet? Let me check to see if I still have a pulse -- yep! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-4

Re: [asterisk-users] Codec choice

Steve, Good point my man...You drinking yet? LOL...I had forgotten about the GROUP and GROUP_COUNT functions, that is a much better way (in that it already existed and doesn't require me to write more code :] ) Slainte! On Fri, Aug 20, 2010 at 7:37 PM, Steve Edwards wrote: > On Fri, 20 Aug 2010

Re: [asterisk-users] Codec choice

On Fri, 20 Aug 2010, Sherwood McGowan wrote: > 1. Set up a Global Variable that will store that kit's current number of calls > 2. Check that variable when a call starts (but before you dial out) > 3. If the number of calls is <49 (since the current call will make > 50), use codec A via the CHANNE

Re: [asterisk-users] Codec choice

1. Set up a Global Variable that will store that kit's current number of calls 2. Check that variable when a call starts (but before you dial out) 3. If the number of calls is <49 (since the current call will make 50), use codec A via the CHANNEL() function, otherwise use codec B using the same fun

[asterisk-users] Codec choice

Hi, Thanks. Actually can it be done on whole kit basis rather than for an extension or peer. Like if there are lot of inbound sip interconnects on a kit , how can we send first 50% simultaneous calls to dahdi with codec A and after that with codec B. Thanks, D -- __

Re: [asterisk-users] Codec choice

> On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan >> >> Does anyone has an idea how to tell asterisk to use codec A for first >> 50 calls and then codec B for rest of the calls. On Thu, 19 Aug 2010, Sherwood McGowan wrote: > the easiest way I can think of is to use a global variable that you

[asterisk-users] Codec choice

Ok. And how will we do for getting sip inbound calls from different ips and sending them to dahdi. Thanks, D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us fo

Re: [asterisk-users] Codec choice

i do this by having 2 peers setup, one has a call limit of 10 and uses g729, the rest of the calls get sent to the second peer which uses ulaw. all calls attempt peer 1 if there's channels available it uses it if not it just moves through the dialplan to use the second one. On 19 August 2010 09:1

Re: [asterisk-users] Codec choice

On Thu, Aug 19, 2010 at 3:37 AM, Steve Howes wrote: > > On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: >> Does anyone has an idea how to tell asterisk to use codec A for first 50 >> calls and then codec B for rest of the calls. > > You could create two separate trunks, one for each codec? > >

Re: [asterisk-users] Codec choice

On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. You could create two separate trunks, one for each codec? S -- __

Re: [asterisk-users] Codec choice

On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan wrote: > Hi, > > > > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. > > > > Thanks, > > Deepika > > -- > _

[asterisk-users] Codec choice

Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Codec Conversion

El 09/08/10 05:30, michel freiha escribió: Hello Miguel molina, I did what you asked, but still the voice is too bad Regards On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina mailto:mmol...@millenium.com.co>> wrote: El 05/08/10 14:50, Tim Nelson escribió: - "michel freiha"

Re: [asterisk-users] Codec Conversion

Hello Miguel molina, I did what you asked, but still the voice is too bad Regards On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina wrote: > El 05/08/10 14:50, Tim Nelson escribió: > > - "michel freiha" wrote: > > > > Dear Sir, > > > > I tried to convert ilbc to ulaw and get the same result

Re: [asterisk-users] Codec Conversion

Steve- > On 08/07/2010 03:15 AM, Jeff Brower wrote: >> Steve- >> >>> El 05/08/10 14:50, Tim Nelson escribió: - "michel freiha"wrote: > Dear Sir, > > I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality > Regard

Re: [asterisk-users] Codec Conversion

On 08/07/2010 03:15 AM, Jeff Brower wrote: > Steve- > >> El 05/08/10 14:50, Tim Nelson escribió: >>> - "michel freiha"wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice >>> Quality Regards >>>

Re: [asterisk-users] Codec Conversion

Steve- > El 05/08/10 14:50, Tim Nelson escribió: >> - "michel freiha" wrote: >>> Dear Sir, >>> >>> I tried to convert ilbc to ulaw and get the same result...Bad Voice >> Quality >>> Regards >>> >> Again, iLBC is poor quality to begin with. You can't take

Re: [asterisk-users] Codec Conversion

On 08/06/2010 04:43 PM, Jeff Brower wrote: > Steve- > >>On 08/06/2010 05:40 AM, Jeff Brower wrote: >>> Miguel- >>> El 05/08/10 14:50, Tim Nelson escribió: > - "michel freiha" wrote: >> Dear Sir, >> >> I tried to convert ilbc to ulaw and get the same result...Bad Vo

Re: [asterisk-users] Codec Conversion

On Fri, 06 Aug 2010 07:40:44 -0500, Michael Graves wrote: >On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote: > > > >>> MELPe is patent encumbered, >> >>Not if used for govt/defense purposes. For commercial-only purposes, TI will >>waive royalty fees if their chip is used >>in the produ

Re: [asterisk-users] Codec Conversion

On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote: >> MELPe is patent encumbered, > >Not if used for govt/defense purposes. For commercial-only purposes, TI will >waive royalty fees if their chip is used >in the product. It would have been nice if Digium had considered the many >adv

Re: [asterisk-users] Codec Conversion

Steve- > On 08/06/2010 05:40 AM, Jeff Brower wrote: >> Miguel- >> >>> El 05/08/10 14:50, Tim Nelson escribió: - "michel freiha" wrote: > Dear Sir, > > I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality > Regards > Again, iLBC is p

Re: [asterisk-users] Codec Conversion

On 08/06/2010 05:40 AM, Jeff Brower wrote: > Miguel- > >> El 05/08/10 14:50, Tim Nelson escribió: >>> - "michel freiha" wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice >>> Quality Regards >>> Again, iLBC is poor quality to begin

Re: [asterisk-users] Codec Conversion

>> This just made me remember some comment on the iax.conf sample file... >> >> disallow=lpc10; Icky sound quality... Mr. Roboto. >> > LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job > with pitch detection so it tends to have a > 'robotic' sound. With

Re: [asterisk-users] Codec Conversion

Miguel- > El 05/08/10 14:50, Tim Nelson escribió: >> - "michel freiha" wrote: >> > >> > Dear Sir, >> > >> > I tried to convert ilbc to ulaw and get the same result...Bad Voice >> Quality >> > >> > Regards >> > >> >> Again, iLBC is poor quality to begin with. You can't take a poor audio >> sa

Re: [asterisk-users] Codec Conversion

El 05/08/10 14:50, Tim Nelson escribió: - "michel freiha" wrote: > > Dear Sir, > > I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality > > Regards > Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it t

Re: [asterisk-users] Codec Conversion

Michel- > I tried to convert ilbc to ulaw and get the same > result...Bad Voice Quality I think you have to be more specific when you say "bad voice quality". Like what? Worse than a cellphone call? Gaps of audio missing? Robotic or "cyborg" sound? Static? A background tone or buzzing? i

Re: [asterisk-users] Codec Conversion

- "michel freiha" wrote: > > Dear Sir, > > I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality > > Regards > Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'.

Re: [asterisk-users] Codec Conversion

Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson wrote: > - "michel freiha" wrote: > > > > Dear All, > > > > i would like to ask please if someone tried to make a codec conversion > from ilbc to g729,

Re: [asterisk-users] Codec Conversion

- "michel freiha" wrote: > > Dear All, > > i would like to ask please if someone tried to make a codec conversion from > ilbc to g729, because i did that but the voice quality was too bad and a lot > of disconnection.. > > Can i get your feedback regarding this issue please? > > rega

Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

> Only when I configure my Grandstream to use only G726 (I have 8 > choices), I see that the g726-codec is used. > When I configure 7 x g726 and 1 x alaw, then again alaw is used ! > > Is it normal that Asterisk has such a great preference for alaw ?! The > moment the peer suggests codec alaw (ev

[asterisk-users] Codec Conversion

Dear All, i would like to ask please if someone tried to make a codec conversion from ilbc to g729, because i did that but the voice quality was too bad and a lot of disconnection.. Can i get your feedback regarding this issue please? regards -- _

Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

On 08/03/2010 04:21 PM, Philipp von Klitzing wrote: > Also: > > There are at least two implementations of the g726 codec, i.e. g726 and > g726aal2. For this also look at the g726nonstandard setting in sip.conf. > It is quite possible that your problem is here. > I have the following setting in

Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

Hi! > In the [general] section of sip.conf I have : > > disallow=all > allow=g726 > allow=alaw > allow=g729 > allow=gsm So change the order there and see what happens. > > * look at the variable SIP_CODEC for the inbound (first) call leg, and > > in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OU

Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

Also: There are at least two implementations of the g726 codec, i.e. g726 and g726aal2. For this also look at the g726nonstandard setting in sip.conf. It is quite possible that your problem is here. For quick testing to see if the codec works at all: Configure your phones to do g726 only (so n

Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

Hello Philipp, thank you for your answer. On 08/03/2010 01:21 PM, Philipp von Klitzing wrote: >> Question 3 : >> How can I get g726 as first preferred codec ?? >> > Which Asterisk version are you using? > Using Asterisk 1.4.30 > * check if you have disallow/allow settings in the [gen

Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

Hi! > Question 1 : > [Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer - > audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726) > why is combined alaw|g726 and not g726|alaw (reverse) ?? Guess: Here the order presented has no meaning for the order of codec

[asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers a

Re: [asterisk-users] Codec negotiation

On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing wrote: > Hi! > >> Because the codec is already chosen before the call is made, and you >> told it that g722 is permitted. >> >> There are all sorts of discussions in play about codec negotiation, >> but at this point in time, if you want differ

Re: [asterisk-users] Codec negotiation

Hi! > Does the 1.4.26.2-patch also work with asterisk 1.4.30 ?? Most probably - who on this list would you like to test it for you? ;-> Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] Codec negotiation

Hi! > Because the codec is already chosen before the call is made, and you > told it that g722 is permitted. > > There are all sorts of discussions in play about codec negotiation, > but at this point in time, if you want different behaviour you'll need to > go and code it yourself Look at the l

Re: [asterisk-users] Codec negotiation

Does the 1.4.26.2-patch also work with asterisk 1.4.30 ?? I have reported a codec-issue, but there is no solution. Will this patch also answer my question ?? https://issues.asterisk.org/view.php?id=17020 Jonas. On 06/29/2010 09:42 PM, Mindaugas Kezys wrote: Try this: http://www.b2bua.org/w

Re: [asterisk-users] Codec negotiation

; Sent: Tuesday, June 29, 2010 7:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec negotiation > > On 26 June 2010 22:08, Ryan Wagoner wrote: > > I have Polycom phones that support the g722 codec. Adding allow=g722 > > to t

Re: [asterisk-users] Codec negotiation

...@lists.digium.com] On Behalf Of Steve Davies Sent: Tuesday, June 29, 2010 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Codec negotiation On 26 June 2010 22:08, Ryan Wagoner wrote: > I have Polycom phones that support the g722 codec. Adding allow=g

Re: [asterisk-users] Codec negotiation

On 26 June 2010 22:08, Ryan Wagoner wrote: > I have Polycom phones that support the g722 codec. Adding allow=g722 > to the [general] section of sip.conf works great and I can make calls > between the phones using g722. However Asterisk is negotiating g722 > for calls going out my voip provider and

[asterisk-users] Codec negotiation

I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the prov

Re: [asterisk-users] Codec G.129 A vs A/B

On 06/04/2010 02:27 AM, Kyle Kienapfel wrote: > http://en.wikipedia.org/wiki/G.729 > Looks like theres A and B and no "A/B" so theres nothing to worry about > What's the point of quoting a page, if you are not actually going to read it? > On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed

Re: [asterisk-users] Codec G.129 A vs A/B

http://en.wikipedia.org/wiki/G.729 Looks like theres A and B and no "A/B" so theres nothing to worry about On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed wrote: > Dear all, I've read that Asterisk supports only the G.729 A audio > codec. I have several Grandstream IP phones with G.729 A/B

[asterisk-users] Codec G.129 A vs A/B

Dear all, I've read that Asterisk supports only the G.729 A audio codec. I have several Grandstream IP phones with G.729 A/B codec implementation. Does G.729 A/B mean both version A and version B, or A/B is a new version different from A and B and it's not supported by Asterisk ??? Thanks a lot

Re: [asterisk-users] Codec preference

I would also add the following : sip.conf has : [general] disallow=all allow=g729 allow=alaw allow=gsm And again the same in the sip peer definition : disallow=all allow=g729 allow=alaw allow=gsm sip debug shows : [Mar 12 15:28:23] Found audio description format G729 for ID 18 [Mar 12 15:28:

Re: [asterisk-users] Codec preference

If I have this is sip.conf : [general] disallow=all allow=g729 allow=alaw The prefered codecs set in my Grandstream phone is G729, alaw. In the sip peer definition I have commented out 'disallow=' and 'allow='. The prefered codecs set in the Zoiper softphone is alaw, gsm. In the sip peer defin

Re: [asterisk-users] Codec preference

Sip.conf : [general] ;context=default allowguest=no allowoverlap=no allowtransfer=yes realm=mydomain bindport=5060 bindaddr=X.X.X.X maxexpiry=1800 minexpiry=60 mohinterpret=default mohsuggest=default language=be useragent=mycorp dtmfmode = rfc2833 alwaysauthreject = yes ;contactdeny=0.0.0.0/0.

Re: [asterisk-users] Codec preference

Post your Asterisk's sip.conf On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens wrote: > How can I set the prefered codec between 2 calling parties ?? > > My Grandstream supports *G729, alaw and gsm*... in this order. > The Zoiper softphone has *alaw and gsm* as codecs... in that order. > > Althou

[asterisk-users] Codec preference

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI show

Re: [asterisk-users] Codec translation in Asterisk

Nobody to take this one! Am I missing anything in knowing following issue? --Hi Group, --Can anybody explain me in detail how the codec translation happens on --asterisk side when 2 endpoints have different codecs? --Thanking you in advance. SM -- ___

[asterisk-users] Codec translation in Asterisk

Hi Group, Can anybody explain me in detail how the codec translation happens on asterisk side when 2 endpoints have different codecs? Thanking you in advance. --SM -- _ -- Bandwidth and Colocation Provided by http://www.api-di

Re: [asterisk-users] codec conversion

On Tue, 2 Feb 2010, Steve Edwards wrote: > On Tue, 2 Feb 2010, wassim darwich wrote: > >> Thanks for?your reply,ill give?you my situation, iam using my asterisk box >> as a switch ,so my client is sending me ulaw and my voip provider?only >> accept g723 ,So what i have to do is to receive?g711

Re: [asterisk-users] codec conversion

On Tue, 2 Feb 2010, wassim darwich wrote: Thanks for?your reply,ill give?you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider?only accept g723 ,So what i have to do is to receive?g711?codec and convert them to g723 at?asterisk ,i tried t

[asterisk-users] codec conversion

Hi: Thanks for your reply,ill give you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider only accept g723 ,So what i have to do is to receive g711 codec and convert them to g723 at asterisk ,i tried this before but i saw the cpu usage is ov

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