Archives : June-2016
Im playing with the optional URL parameter of the Dial() command, which will also be sent to the called party upon successful connection, if the channel technology supports the sending of URLs in this way.[1]Basically, the following asterisk dialp..
I have a business need to block any 180 Ringing packets coming from an FXOgateway, Grandstream 4108. I use Asterisk 13.9.1, and PJSIP. Is this even possible? I all I may do is hack the source code,will it be PJSIP code or Asterisk code? Any help..
I get this message very often:[Jun 26 23:52:49] ERROR[10396]: channel.c:1278 ast_channel_by_name_cb: BUG!Must supply a channel name or partial name to match!I could file a bug if somebody tells me how to obtain a trace for this. Icannot imagine..
What options do I have to setup Distributed Device State across to multiple Asterisk Servers?If an agent is on the phone on a queue on one of the Asterisk server, other servers will need to about it and therefore, will be able to operate adequate..
I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22But this patch is for the SIP channel driver not PJSIP, right?Is it even possi..
i want debug only app_queue(asterisk 13.9)i have this configuration[general][logfiles]console => notice,warning,error messages => notice,warning,error;full => notice,warning,error,debug,verbose debug => debugsyslog.local1 => warning,errorbut afterasterisk*C..
Hello.Recently Ive installed Asterisk 11.6-cert13 on a physical server. Downloaded g729 codec from asterisk.hosting.lv for version 11 codec_g729-ast110-gcc4-glibc-x86_64-core2-sse4.so and installed as instructed.Asterisk starts and `core show code..
list.When i have an incoming call with attended transfer, im getting two separate records in my CDR with 2 different uniqueids. Do I have any way topostpone my uniqueid from one call to another?Im using SIP on Asterisk 11.19 right now.WBR,Dorof..
all,My provider proxy expects authentication header on BYE packets as well. Is there a way in asterisk to add this header on BYE packets?When proxy replies with a 401 on BYE, asterisk just retransmits the BYEpacket.Rega..
I see that you can configure RLS in pjsip.conf, but does this work with realtime?The wiki refers to pjsip.conf for configuration, but since many of the other items can be in the the DB, I was wondering if RLS ca..