Category : Asterisk Users
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I am working on a project that uses Asterisk ARI ExternalMedia request to stream the RTP audio from Asterisk to an UDP/RTP receiver project.Using slin16 format.1) I believe I am seeing is a 12-byte header followed by 640 bytes of data.Is this correct..
Hi Im attempting to use ICE to be able to present all possible RTP transports to peers. 16.28.0~dfsg-0+deb11u2 (I know its old, but unfortunately Asterisk was removed from debian stable and the version in sid is just broken (opus + voicemail dont w..
Thanks to those on IRC confirming quickly that this was not something supported (yet) in Asterisk.Below is a quick fix/patch to tcptls.c for Asterisk 18 against this particular provider.Dwstatic int check_tcptls_cert_name(ASN1_STRING *cert_str, co..
Not sure where to mention this.Very minor/trivial issue.Just wanted to let someone know.If you go to docs.asterisk.org and the Asterisk REST Interface (at least in both 18 and 20 versions).Go to the Channels. There is a list of Method and path lin..
I am using asterisk 18.14.0 and chan_sip. confbridge has dsp_drop_silence=yes The conf joins all the endpoints in a one-way conf.60+ devices and packets choppy or dropping audio.The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHzWhat e..
Cant you just extend the debug and add further logging to understand the choices being made and why?I believe Ive stated this once or twice when youve brought this issue up on IRC but rewrite_contact has no influence or impact on this. It rewrites incom..
I am looking for a decent provider of SIP Trunks but it has to pass the Stir Shaken token to the next carrier. Does anybody know about any? Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 /0013000000G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:..
I used to use the local channel to create a global variable(dialplan)[default]exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}) to that end, I modified cli.conf[startup_commands]originate local/s extension s@default = yes ..
I tested this issue with version 13 and version 18.In res_odbc.conf, if I add a second, new data source like [asterisk]enabled=yesdsn=asterisksanitysql => select 1isolation => read_committedusername=root;password pre-connect => yesforcecommit => yesconnect_time..
I dont recall seeing the rtp and jitter entries being logged regularly at other customer sites, so I am probably missing something obvious.This site is running asterisk 18.17.1. I enabled debugging to try to help track down an issue.The problem is ..