For a few days now Im 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 messa..
When the T.38 re-INVITE is rejected by your SIP peer, they sent an SDP offer with the audio stream set to port number zero (’0′). This means the audio stream is not active, and thus cannot be used. Hence you could enable the fallback (f) opt..
On 03/02/2012 03:53 AM, Waldemar Gessler wrote: > everyone, > > we have made tests with Asterisk 1.6 and SQL Server (MS SQL) 2008 R2 as > DB for: There is no Asterisk 1.6. there are 1.6.0, 1.6.1 and 1.6.2 release branches. Which version are you usi..
Using AGI script to accept the input from caller but input value is not getting stored in variable. Extract from AGI Script: $agi = new AGI(); $agi-> exec(Background,press_one0&press_two0&press_zero0); $agi-> exec(Read,NUMBER,,1,3); $agi-> verbose (..
We arerunning FreeTDM on a very cheap Atcom card, which used another module ph_tor3_e1 on top of Dahdi. I believe this is derived from Torrenta. http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c On Ubuntu 10.04 this gives prob..
all, It disturbs me to see asterisk (v 126.96.36.199) writing CDRs even when there are 0 active channels and 0 active calls. Is there an upper limit in terms of CDRs / second that asterisk can handle? Does it queue the unwritten CDRs somewhere? Please h..