search for: mytrunk

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:35] DTMF[4238] chan...

...cted to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to 'ringythingy.dev1ce.com' -- Executing [blah at anveo_sip:1] Dial("PJSIP/demo-alice-00000005", "PJSIP/blah at mytrunk") in new stack -- Called PJSIP/blah at mytrunk -- PJSIP/mytrunk-00000006 is ringing -- PJSIP/mytrunk-00000006 is ringing -- PJSIP/mytrunk-00000006 is making progress passing it to PJSIP/demo-alice-00000005 > 0x7ff39839e360 -- Strict RTP learning after remote address s...

...13 Phone1 transfers call in Asterisk 13 to Phone 2 But the transfer fails with an "NOTIFY 400 Bad Request". In Asterisk log I don't see any reference to "external_replaces" extension when the REFER arrives pjsip.conf [simpletrans] type=transport protocol=udp bind=0.0.0.0 [mytrunk] type=registration transport=simpletrans outbound_auth=mytrunk server_uri=sip:111 at 89.1.23.217:5060 client_uri=sip:111 at 89.1.23.217:5060 [mytrunk] type=auth auth_type=userpass password=111 username=111 [mytrunk] type=aor contact=sip:89.1.23.217:5060 [mytrunk] type=endpoint transport=simpletr...

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 can't change anything there. I don't see any examples here

...n POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want to send no more than 12 channels exten => _0692XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) exten => _0693XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) How would you go about it? Currently my IAX2 peer definition looks like this: # machine in london [mytrunk] type=friend...

...de95ec89 currently running on dunkel > (pid = 517890) > dunkel*CLI> > dunkel*CLI> > == Setting global variable 'SIPDOMAIN' to 'ringythingy.dev1ce.com' > -- Executing [blah at anveo_sip:1] Dial("PJSIP/demo-alice-00000005", > "PJSIP/blah at mytrunk") in new stack > -- Called PJSIP/blah at mytrunk > -- PJSIP/mytrunk-00000006 is ringing > -- PJSIP/mytrunk-00000006 is ringing > -- PJSIP/mytrunk-00000006 is making progress passing it to > PJSIP/demo-alice-00000005 > > 0x7ff39839e360 -- Strict RTP l...

...requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan should jump to the next priority. exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) exten => 1001,2,,NoOP(${DIALSTATUS}) exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1) exten => h,1,NoOp() exten => h,n,NoOP(${DIALSTATUS}) ----------------------------------------------------------------------- If i try to dial the same offline endpoint with the below code snippet, it jumps to next prirorty. exten => 1001,1,Dial(PJSIP/${EXTEN}) exten...

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volg...

...unkel.dev1ce.com>;tag=3166828162 To: <sip:13107950860 at dunkel.dev1ce.com> CSeq: 8613 INVITE Server: Asterisk PBX GIT-master-0cde95ec89 Content-Length: 0 -- Executing [13107950860 at anveo_sip:1] Dial("PJSIP/demo-alice-00000002", "PJSIP/13107950860 at mytrunk") in new stack -- Called PJSIP/13107950860 at mytrunk == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'PJSIP/demo-alice-00000002' status is 'CONGESTION' <--- Transmitting SIP response (548 bytes) to TCP:[2...

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp. I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk definition. It does not seem that anything is listening on 5068? How can I run SIP tcp on port 5068? telnet localhost 5068 Trying 127.0.0.1... telnet: connect to address 127.0.0.1:

...ic ip ? volga629 On Sun, 3 Dec, 2017 at 11:01 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:55 AM, volga629 at networklab.ca wrote: >> If understand correctly type=identify is more for sip trunk >> configuration ? >> >> >> ;[mytrunk] >> ;type=identify >> ;endpoint=mytrunk >> ;match=198.51.100.1 >> ;match=198.51.100.2 >> >> >> In chan_sip it was just reply 200 OK on keepalive packet without >> need >> define trunks. >> >> > > All incoming traff...

Right now it reply 401 Unauthorized with message in log "No matching endpoint ..." on Content 0 should reply 200 OK I guess <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> OPTIONS sip:10.30.100.27:5080 SIP/2.0 Via: SIP/2.0/UDP 10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0 To: <sip:10.30.100.27:5080> From: <sip:vprx00 at

...[my_provider] type=identify endpoint=my_provider match=sip.example.com And it registers fine: <Registration/ServerURI..............................> <Auth..........> <Status.......> ========================================================================================== mytrunk/sip:sip.example.com my_provider Registered And when it gets an INVITE from my provider (192.168.0.1): <--- Received SIP request (997 bytes) from UDP:192.168.0.1:5060 ---> INVITE sip:1235551212 at 10.75.22.5:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5...

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian