similar to: Asterisk not using common codec between (SIP) endpoints

Displaying 20 results from an estimated 900 matches similar to: "Asterisk not using common codec between (SIP) endpoints"

2012 Nov 21
1
core show translation - difference in Asterisk Versions
Hello All, I was wondering if somebody could elaborate the change in translation of codecs specifically the amount of time increased in Asterisk 11. For example *Asterisk 11* * **alaw **speex * *gsm **15000 **15000 * *ulaw 9150 15000* * * *Asterisk 1.6.x* * **alaw **speex * *gsm **2 12002 * *ulaw 1 12002* I did recalculate the
2010 Feb 08
3
High codec translation times on x64
Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723
2014 Feb 11
0
g726 transcoding
Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"
2012 Mar 21
0
AMR Codec with Asterisk 1.8.9.1
Hi All, I would like to configure AMR codec in asterisk 1.8.9.1. After lots of research i found " http://sourceforge.net/projects/asterisk-amr/files/" thie link, and follow steps to configure amr. codec_amr.so successfully compiled and load in asterisk. *> core show translation * Translation times between formats (in microseconds) for one second of data Source
2010 Aug 20
2
codec_g729.so not work!
hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin
2019 Jul 05
2
Asterisk and Linphone
I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000
2011 Oct 11
0
Asterisk 1.8.7 and VoiceMailMain
Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read
2020 Jun 08
0
pjsip extensions rings but call drop on answer
Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing
2011 Sep 30
1
Core show translation > 4000ms
Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME
2013 Feb 28
1
Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and
2014 Dec 11
0
PJSIP configuration question
I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >
2006 Dec 15
2
call from h323 to SIP
Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =
2008 Feb 08
0
Transcoded G.722 calls unintelligible with recent SVN head
For about 10 months I have been running a succession of Asterisk SVN trunk versions on an Athlon 64 X2 4400+ based machine with OpenSuSE 10.2 at my home. I have a variety of SIP phones (mostly Polycom) internally; my external connections are two POTS lines on a TDM400P (with HPEC) and an IAX2 link to a VoIP provider. I had Asterisk configured to allow G.722 and u-law on the Polycom phones,
2016 Dec 10
6
failing to start asterisk on centos7
ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under
2009 Oct 13
3
strange transcoding values
Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000
2020 Jun 13
0
Voice "broken" during calls
So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon