Archives : November-2017
Im running Asterisk 15.1.0 and in the process of converting my various SIP endpoints to use PJSIP.My Zoiper client causes the messages quoted below to show up on the CLI once per minute.Things seem to work OK, but I am curious because there seems..
Now that Joshua had the kindness to respond, I see here a big disconnect between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like me would need to avoid proxying the media.Would would Digium support and bring in with such fanfar..
Please correct me if I am wrong. With PJSIP there is no way for Asterisk to stay a OUT of the media path, while with the old SIP channel, using directrtpsetup and directmedia, it just works. The issue I think is that other servers do not accept reinvi..
I tried many times and this dial model fails for me SIP/g729-outbound/15555555555/192.168.1.120The peer g729-outbound does exist but it does not have a host line, that is why I am supplying the host dynamically for each call. According Asterisk13 f..
[sub-out-do-dial] exten => s,1,NoOp(Dial) same => n,NoOp(FirstChannel: ${CHANNEL}) same => n,Dial(????,60,gF) same => n,NoOp(SecondChannel: ${CHANNEL}) same => n,Return() [some] exten => s,1,GoSub(sub-out-do-dial,s,1) In case of the destinat..
All;I have a customer who is looking for a particular DID. (I dialed it and it is not in service). I searched through my preferred upstream providers list but I came up empty. I wrote them, and this is their reply. We currently do not have that speci..
Hello! Looks like faq, but… Could you , please, point me on how to convert this [cisco] type=friend host=192.168.22.253 insecure=port,invite to pjsip? as you can see another side is very old cisco router, so I cant change anything there. I dont ..
Hello!Im facing the following scenario:- Initial call opened to asterisk: SDP g722,alaw,ulaw- Outgoing call to provider started with Invite / SDP alaw, g726 andg729.- Provider sends 183 Session progress SDP: g729, alaw- Provider sends g729 rtp packagesB..