Re: [asterisk-users] 100% CPU after upgrade. (Solved)

I had meant to post a follow up to this, but just... didn't. Sorry. Anyway, I had made a silly change to my safe_asterisk script that caused it to start asterisk in the background, but also with a console. This caused asterisk to try to write to a non-existent console tty. Dumb mistake on my

Re: [asterisk-users] ** in extensions.conf

Is ** also defined in features.conf? Thanks; John -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, April 26, 2017 05:41 PM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] SIP and Voice on different nets

​I have connection with two networks (by VoIP provider setup) 1 - 10.10.10.0/24 = SIP 2 - 10.10.11.0/24 = Voice How to tell Asterisk send / receive voice traffic not on SIP network. When I look into dumps, I see Asterisk trying to use SIP net for voice Unfortunately, I _need_ to use two networks

Re: [asterisk-users] ** in extensions.conf

On Wed, 26 Apr 2017 at 20:29 Jerry Geis wrote: > I just tried this in my extensions.conf > > exten => **,1,Noop(Testing) > exten => **,n,Playback(demo-congrats) > > Did a reload... and the above does not happen. > I created as 12 instead of the ** and that works fine. > >

Re: [asterisk-users] SIP and Voice on different nets

On Thu, Apr 27, 2017, at 09:10 AM, Artem Chekulaev wrote: > ​I have connection with two networks (by VoIP provider setup) > 1 - 10.10.10.0/24 = SIP > 2 - 10.10.11.0/24 = Voice > > How to tell Asterisk send / receive voice traffic not on SIP network. > When > I look into dumps, I see Asterisk

Re: [asterisk-users] SIP and Voice on different nets

Yes, Voice = RTP Using chan_sip 2017-04-27 15:32 GMT+03:00 Dovid Bender : > By voice do you mean RTP? Are you using chan_sip or pjsip? > > > On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev > wrote: > >> ​I have connection with two networks (by VoIP

Re: [asterisk-users] SIP and Voice on different nets

By voice do you mean RTP? Are you using chan_sip or pjsip? On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev wrote: > ​I have connection with two networks (by VoIP provider setup) > 1 - 10.10.10.0/24 = SIP > 2 - 10.10.11.0/24 = Voice > > How to tell Asterisk send / receive

Re: [asterisk-users] SIP and Voice on different nets

Seems I responded the same time as Josh. Follow what he has suggested. On Thu, Apr 27, 2017 at 8:41 AM, Artem Chekulaev wrote: > Yes, Voice = RTP > > Using chan_sip > > 2017-04-27 15:32 GMT+03:00 Dovid Bender : > >> By voice do you mean RTP? Are you

Re: [asterisk-users] TDM400P takes too long to ring

I am getting caller id fine on the phones and console, but not sure about the formatting your are talking about. It always just worked for me in the past. Is there something I can easily see to know if I’m not setting it right? From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] TDM400P takes too long to ring

If you are trying to detect caller ID, and it is being supplied by the telco in the format you have configured in /etc/chan_dahdi.conf then this should not cause a delay. Are you actually seeing the caller ID being displayed on the ringing phones? If, however, the telco is not supplying caller ID

[asterisk-users] TDM400P takes too long to ring

Hey all, I have a setup with two analog lines coming into and Asterisk 13 box with a TDM400P and it takes a lot of rings before asterisk takes over. I've traced this same box on two different analog providers so it probably isn't a problem with them. I DO have callerid enabled and not sure I