Re: [asterisk-users] No compatible codecs, not accepting this offer!

2014-01-15 Thread Francesco Namuri
Il 15/01/2014 09.59, Francesco Namuri ha scritto: > Hello, > I'm having this issue on my pbx, it appears that asterisk is refusing > the codecs that my providers is proposing. > My trunk configuration is: > > --- > username=5x5x7x9x0x3 > type=friend > secret=CRcxn7sqwm > qualify=yes > port=5060 > i

Re: [asterisk-users] No compatible codecs, not accepting this offer!

2014-01-15 Thread James Sharp
On 1/15/2014 5:50 AM, Gareth Blades wrote: On 15/01/14 09:39, Francesco Namuri wrote: Hello James, thanks for your answer, I supposed this too, but my provider answered me that as m=audio 43718 RTP/AVP 8 18 3 101 ^ ^ ^ GSM proposal

Re: [asterisk-users] No compatible codecs, not accepting this offer!

2014-01-15 Thread Gareth Blades
On 15/01/14 09:39, Francesco Namuri wrote: Hello James, thanks for your answer, I supposed this too, but my provider answered me that as m=audio 43718 RTP/AVP 8 18 3 101 ^ ^ ^ GSM proposal ^ ^--

Re: [asterisk-users] No compatible codecs, not accepting this offer!

2014-01-15 Thread Francesco Namuri
Il 15/01/2014 10.09, James Sharp ha scritto: > On 1/15/2014 3:59 AM, Francesco Namuri wrote: >> Hello, >> I'm having this issue on my pbx, it appears that asterisk is refusing >> the codecs that my providers is proposing. >> My trunk configuration is: >> > > Pretty simple - > > >> --- >> username=5

Re: [asterisk-users] No compatible codecs, not accepting this offer!

2014-01-15 Thread James Sharp
On 1/15/2014 3:59 AM, Francesco Namuri wrote: Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: Pretty simple - --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insec

[asterisk-users] No compatible codecs, not accepting this offer!

2014-01-15 Thread Francesco Namuri
Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insecure=port,invite host=sip.txtxlxoxp.it fromuser=5x5x7x9x0x3 fromdomain

Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Ricardo Carvalho
Problem SOLVED. You'r right, this is a problem of codec mismatching. Activating sip debug i can see it: Capabilities: us - 0x802 (gsm|g726), peer - audio=0x10d (g723|ulaw|alaw|g729) [May 9 17:16:37] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! I solv

Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Andres
On 5/9/2012 11:56 AM, Ricardo Carvalho wrote: That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. My guess is the incoming call is not being matched with the peer you are expecting. Do a sip debug and watch the output to see what pe

Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Eric Wieling
risk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11 That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw an

Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Ricardo Carvalho
That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw and alaw shows up when i do "core show codecs audio" in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/mo

Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread A J Stiles
On Wednesday 09 May 2012, Ricardo Carvalho wrote: > [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible > codecs, not accepting this offer! > > Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tre

[asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Ricardo Carvalho
Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore.

Re: [asterisk-users] No compatible codecs, not accepting this offer

2012-03-04 Thread SamyGo
HI, Instead of taking traces on asterisk console, try capturing traffic in pcap using tcpdump command and later analyze it with wireshark. #tcpdump -i eth*N *-s 0 -w sip-capture.pcap -v Download this file to any machine with wireshark installed and apply "sip" as filter there. Regards. Sammy On F

[asterisk-users] No compatible codecs, not accepting this offer

2012-03-02 Thread Markus
For a few days now I'm getting "chan_sip.c: No compatible codecs, not accepting this offer!" on the CLI and in the messages log in irregular intervals. How can I find out what (who) is causing this? If I turn on "sip debug" I get flooded with SIP messages from peers/users (qualify=yes etc.) an

[asterisk-users] No compatible codecs / static noise

2008-04-11 Thread Joseph
I'm running asterisk 1.2 with Sipura adapters. I've tried to experiment with different codes but I'm either getting "No compatible codecs" if I use "gsm" or static noise if I use "g726" I was under impression that asterisk would translate between codecs according to "show translation" table. 2

Re: [asterisk-users] No compatible codecs!

2008-02-19 Thread Louwrens Benadé
screen, you should see what is missing. Good luck. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Naveen Palani Sent: 19 February 2008 02:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No compatible codecs! I read from the forums, that if i

Re: [asterisk-users] No compatible codecs!

2008-02-19 Thread Naveen Palani
I read from the forums, that if i build mysql the problem will be resolved. As i get the similar warning message for MeetMe(). How to build mysql or MeetMe manually?? Regards, Naveen.Palani - Original Message - From: Naveen Palani To: asterisk-users@lists.digi

Re: [asterisk-users] No compatible codecs!

2008-02-19 Thread Naveen Palani
Resolved! The problem was with sip.conf file. I had to comment the lines allow=alaw allow=ulaw This made the trick.. I am trying to get the mysql database connection from my asterisk box. Installed the asterisk-addons-1.4.4 version on the same box where asterisk and mysql is installed. When

Re: [asterisk-users] No compatible codecs!

2008-02-18 Thread Alex Balashov
Naveen Palani wrote: > Hi, > > I have the asterisk-1.4.11 set up installation on my Ubuntu machine. > When i try making a simple incoming call using xlite softphone. I get > the following message when i try calling to the number. > > *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 proce

[asterisk-users] No compatible codecs!

2008-02-18 Thread Naveen Palani
Hi, I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try making a simple incoming call using xlite softphone. I get the following message when i try calling to the number. *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No compatible codecs, not ac

[Asterisk-Users] No compatible codecs!

2005-03-24 Thread Gouri Johannsen
Hello, I have been having this problem for several releases of Asterisk.  Whenever, I use iLBC or Speex codecs to make a SIP call, I get "No compatible codecs!" error, even though I am not disallowing anything in my sip.conf.  One way I made it work is to hard-code these codecs in global_capabil

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-02 Thread Eric Wieling aka ManxPower
Philipp von Klitzing wrote: Hi Eric! and do NOT use bandwidth= Why is that? I am curious... Because bandwidth= just enables specific codecs. The specific codecs enabled depend on the bandwidth= setting. ___ Asterisk-Users mailing list Asterisk-Users@l

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-02 Thread Philipp von Klitzing
Hi Eric! > and do NOT use bandwidth= Why is that? I am curious... Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http:

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-01 Thread Peter Bowyer
On Tue, 01 Mar 2005 16:31:20 -0500, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote: > > >> To say that I'm confused would be understating things rather severely. > > > Kevin P. Fleming wrote: > > > > > To say that we can't help you without seeing your config files would > > also be an understatement. U

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-01 Thread Kevin P. Fleming
Ken D'Ambrosio wrote: Anyway, bottom line: all's well, now. Had to enter some extra stuff into my sip.conf file, but the big clue was the fact that it even -could- have been a config file problem, so many thanks. Yes, sometimes what happens is that you are accidentally using undocumented behavi

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-01 Thread Eric Wieling
Ken D'Ambrosio wrote: Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0 sources that I downloaded from Digium, my Polycom 300 works just fine. If I build with either various CVS builds, or the 1.0.6 sources from Digium, I get "No compatible codecs!". WTF? I'm using -the

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-01 Thread Ken D'Ambrosio
To say that I'm confused would be understating things rather severely. Kevin P. Fleming wrote: To say that we can't help you without seeing your config files would also be an understatement. Unfortunately, we are not all-knowing nor telepathic, so just saying "it doesn't work" won't generate mu

Re: [Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-01 Thread Kevin P. Fleming
Ken D'Ambrosio wrote: To say that I'm confused would be understating things rather severely. To say that we can't help you without seeing your config files would also be an understatement. Unfortunately, we are not all-knowing nor telepathic, so just saying "it doesn't work" won't generate much

[Asterisk-Users] "No compatible codecs!" -- worked with 1.0.0, not 1.0.6 or CVS.

2005-03-01 Thread Ken D'Ambrosio
Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0 sources that I downloaded from Digium, my Polycom 300 works just fine. If I build with either various CVS builds, or the 1.0.6 sources from Digium, I get "No compatible codecs!". WTF? I'm using -the exact same- config fil

Re: [Asterisk-Users] No compatible codecs

2005-01-18 Thread jeffrey johnson
this may help you http://billing.mutualphone.com/phpBB2/viewtopic.php?t=78&postdays=0&postorder=asc&start=15 On Tue, 18 Jan 2005 10:23:45 -0500, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote: > > Original Post > > I have an Asterisk related problem with mutualphone. > I

RE: [Asterisk-Users] No compatible codecs

2005-01-18 Thread Kanuri, Seshu (Company IT)
Original Post I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinatio

Re: [Asterisk-Users] No compatible codecs (solved)

2005-01-17 Thread Rene Kluwen
" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, January 17, 2005 2:04 AM > Subject: Re: [Asterisk-Users] No compatible codecs > > > > I've heard problems with the Grandstream G729 and the new digiu

Re: [Asterisk-Users] No compatible codecs

2005-01-17 Thread Rene Kluwen
AM Subject: Re: [Asterisk-Users] No compatible codecs > I've heard problems with the Grandstream G729 and the new digium G729 > by MAC ID. Could be a compatibility issue with the implementations. > Did you ever use the Grandstream against asterisk with the old > Voi

Re: [Asterisk-Users] No compatible codecs

2005-01-16 Thread William Suffill
I've heard problems with the Grandstream G729 and the new digium G729 by MAC ID. Could be a compatibility issue with the implementations. Did you ever use the Grandstream against asterisk with the old Voiceage G729? I've heard that works just fine. -- William __

Re: [Asterisk-Users] No compatible codecs

2005-01-16 Thread Rene Kluwen
ndres" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, January 16, 2005 11:46 PM Subject: Re: [Asterisk-Users] No compatible codecs > > >Any suggestions about what I can change to make this work? > > > > > Ye

Re: [Asterisk-Users] No compatible codecs

2005-01-16 Thread Andres
Any suggestions about what I can change to make this work? Yes, you should get a G729 license for your Asterisk. Cheers! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To U

[Asterisk-Users] No compatible codecs

2005-01-16 Thread Rene Kluwen
I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call

[Asterisk-Users] No Compatible codecs? Got license

2004-07-12 Thread Walter Klomp
Hi,   I have a Cisco 5300 which I want to make a call THROUGH the Asterisk PBX (security) to an IP phone which supports g729, and vice versa. Both Cisco and the phone talk this codec if I do not force the call to go through *   However if I say canreinvite=no in the sip.conf for either