search for: voipms

...d go where they should. Two problems remain. 1. Still can't register my phone The username and password are correct. I don't know what else to try. 2. Asterisk can't find the extension in my inbound context. [May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite:  voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because extension not found in context 'voipms-inbound'. I changed the name of the context in pjsip's  to 'voipms-inbound' and removed reference to '[mycontext]' from pjsip.conf and extensions.conf as...

On 5/24/23 08:03, Steve Matzura wrote: > > ***  extensions.conf  *** > > > [general] > > [globals] > > ; Make sure to include inbound prior to outbound because the > _NXXNXXXXXX handler will match the incoming call and create a loop > include => voipms-inbound > include => voipms-outbound > > [voipms-outbound] > exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _1NXXNXXXXXX,n,Hangup() > exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms) > exten => _NXXNXXXXXX,n,Hangup() > exten => _011.,1,Di...

..."480 Temporarily Unavailable" and a fast busy. Doing a "sip show peers" appears normal. When I do a detailed "sip show mypeername", the one anomalous thing I see is that that the "Addr->IP" setting is listed as "(Unspecified)". * Name : voipms Secret : <Set> [...] ToHost : dallas.voip.ms Addr->IP : (Unspecified) Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP [...] In contrast, after I do a "sip reload", outbound calls start working again and the "si...

This was supposed to go to the list. I am now thoroughly confused. In the [voipms] stanza where endpoint is defined (type=endpoint), everything points to voipms. But in the [yealink] stanzas, I tried pointing everything to Steve, one item at a time, then both of them, and nothing changed. On 5/24/2023 10:00 AM, Stefan Tichy wrote: block quote Am Wed, May 24, 2023 at 09:40:18...

>Using system() you could issue a asterisk -rx 'core restart now' So I tried this exten => s,1,Playback(beep) exten => s,n,Dial(Console/default,20,g) exten => s,n,Hangup exten => s,n,System(asterisk -rx 'core restart now') But it does not continue. Last thing I see is "Exited non zero" so its not doing the hangup or the system. jerry --------------

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

Asterisk 12.5 I'm using AMI to initiate a "call me now" feature from the web site. The AMI looks like: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same =>n,Queue(sales,Rtc) [dial-to-customer] exten => s,1,NoOp(dial-to-customer channel=${CHANNEL(name)}) same =>n,DumpChan() The dial-to-customer context is invoked when the sa...

...AMI request: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Callerid: Call Me Now <777> Variable: MMCUSTOMER_NUMBER=9995551212 Timeout: 999999 Output from the subroutine: -- Executing [s at dial-to-customer-sub:3] Verbose("SIP/voipms-0000001e", "2, dial-to-customer-sub interface name customer number operatoriod channel name SIP/voipms-0000001e unique ID mlcx500-1375465508.61 ") in new stack The "U" subroutine seems to work OK same =>n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,U(dial-to-customer-sub,...

...out 20 Hz signal (a bit like a robot voice). What could be the problem? (Not using pulseaudio; +- default configuration.) One hypothesis I have is that it uses a too small buffer somewhere. - I don't understand how the extensions stuff is working. voip.ms wiki told me to create sections named [voipms], but how do I switch to 'default'? tie*CLI> console dial 4443 No such extension '4443' in context 'default' tie*CLI> console dial 04443 No such extension '04443' in context 'default' tie*CLI> console dial 004443 No such extension '004443' i...

Doug from this list got me to change my connectivity to my DID provider from SIP to IAX, and bingo, it all just worked instantly. For my next trick: setting up voicemail. The book does it all with smoke and mirrors (SQL), but I'm fresh outa those, so I'll be doing it the old-fashioned way, by editing the voicemail.conf and users.conf files with some hopefully helpful hints from our

...X Subclass: POKE Timestamp: 00014ms SCall: 00890 DCall: 00000 [67.205.74.184:4569] Notice there are no Rx-Frames, and my peer table looks like this: dlaptop*CLI> iax2 show peers Name/Username Host Mask Port Status Description voipms/121322_i 184.75.215.106 (S) 255.255.255.255 4569 UNREACHABLE voipms2/121322_ 67.205.74.184 (S) 255.255.255.255 4569 UNREACHABLE 2222/2222 99.245.204.155 (S) 255.255.255.255 4569 UNREACHABLE...

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start