[asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

Here's a weirdness - I got a call from someone who couldn't get to my info line earlier, I tried it and it was busy tone. Being on a layby beside a road on a mobile on a long journey, my only real option was a remote server reboot so I couldn't diagnose further. That fixed it, but here's the

[asterisk-users] Audio cut-outs

I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card

Re: [asterisk-users] Audio cut-outs

I had this recently... and i bet if you use wireshark/tcpdump youll see a dns lookup for the server's own hostname right before the cutout, and audio again after response is received. quick fix is to add the hosts name and ip to /etc/hosts https://issues.asterisk.org/jira/browse/ASTERISK-26280

Re: [asterisk-users] Dial and start music on hold after timeout

Thank you, I just tried your suggestion. Strangely, the announcement is played only if I try to dial a SIP peer which is not available (not registered to be more precise). If the SIP peer is available, I only get the ring tone, and never hear the announcement. Here is the dialplan (I had to

Re: [asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

On Tue, Aug 23, 2016 at 11:19 AM, Jonathan H wrote: > Here's a weirdness - I got a call from someone who couldn't get to my info > line earlier, I tried it and it was busy tone. > > Being on a layby beside a road on a mobile on a long journey, my only real > option was a

Re: [asterisk-users] Dial and start music on hold after timeout

Damn, I was going to suggest trying a Queue with a single member using the 'r' option to play ringing instead of MOH and using an announcement but the queue will stop ringing your agent while it plays the announcement. It'd go right back to ringing after the announcement however. On Mon, Aug 22,

Re: [asterisk-users] Dial and start music on hold after timeout

Maybe try progress() instead of answer () בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" כתב: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If

Re: [asterisk-users] Asterisk Realtime RTUPDATE issue

Ahmed Munir wrote: Hi, I'm currently using Asterisk 11.7.0.The issue currently I'm facing in Asterisk realtime sip_buddies table i.e. if I try to unregister the extension, ipaddr, port, regseconds, fullcontact, useragent and lastms remain still populated with data unless do the sip reload. This

Re: [asterisk-users] Dial and start music on hold after timeout

You could m and make a moh file that has ringing the first 30 sec and then the anouncment בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" כתב: > Thank you for the idea. The problem with RetryDial, is that it will cancel > the first call, play the announce and then dial the

Re: [asterisk-users] Dial and start music on hold after timeout

How about: exten => s,1,Dial(SIP/alice/555@delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for