Tag : sip
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Hi. Does anyone have any recommendations for a *really* real-time monitoring solution for Asterisk? Im thinking that something like Grafana (which Ive played with for another purpose, but dont really use yet) can do a good job of displaying the d..
Is there a way to configure the old SIP channel to stay in sip set debug all the time, without human intervention and also at boot time, b..
I work on the Asterisk side of things and admit to not knowing about browser development.A co-worker asked me today why they should develop a web based agent software using WebRTC?They prefer to develop using a SIP based javascript library they found…
All,I tried to switch from SIP to PJSIP but I cant make any calls. Asterisk 15.4.0Clients: MicroSIP (based on the pjsip SIP stack)With sip.conf all functions OK (SIP instead of PJSIP in extensions.conf)I converted SIP to PJSIP with the script contrib/scripts/sip_to_pjsip/sip_to_pjsip..
I need some help understanding SIP dialog. Some actor is trying to access my server, but I cant figure out what hes trying to do ,or how.Im getting a lot of these warnings.[May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: Retransmission time..
all,we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become al..
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..
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..
Has anyone used Telynx as a SIP trunk provider? It works with chan_sip but it I seem to be having problems trying to set up a PJSIP trunk. I always get a 401 Unauthorized when they send me a call. I know my username and password are correct si..
I am trying to get the user-agent from extensions registered via pjsip. With sip we could do a sip show peer peername and it would list the user-agent string. In a pjsip deployment it looks like this info is likely in the contact. I know we can acc..