search for: norefersub

Displaying 20 results from an estimated 57 matches for "norefersub".

2019 Mar 25
2
Asterisk Transfers
Does anyone know if there is a way to disable the norefersub for PJSIP? It appears this is causing problems with a test we're running with Cisco. A wireshark trace from a system where the transfer with Cisco works versus a trace with Asterisk/Cisco shows one big difference being the supported: norefersub The REFER Accepted response is received by Aster...
2019 Mar 28
3
Asterisk Transfers
On Thu, Mar 28, 2019, at 11:10 AM, Dan Cropp wrote: > > Is there no one who knows if there is a way to turn off the norefersub setting? > > > Supported: norefersub > > > This happens in the TRYing, OK, and other commands in response to the INVITE. > > > For chan_sip, I noticed it does not send the norefersub. As a result, > Cisco then sends NOTIFY packets with TRYing, Ringing, OK insid...
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
...36146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE Server: FPBX-13.0.188.8(13.11.2) Contact: <sip:80.142.13.32:55060> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 286 v=0 o=- 15363811 15363814 IN IP4 192.168.3.99 s=Asterisk c=IN IP4 80.142.13.32 t=0 0 m=audio 51822 RTP/AVP 8 3 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20...
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57
2007 Apr 12
2
Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
...: <sip:215@10.0.10.136:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Cisco 7940" <sip:215@10.0.10.5>;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136 s=SIP Call t=0 0 m=audio 16946 RTP/AVP 8 0 18 101 c=IN IP4 10.0.10.136 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a...
2009 Aug 17
2
Accessing to ekiga.net through Asterisk
...rom: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org CSeq: 183 INVITE Contact: <sip:201 at 10.1.0.65> Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 2122879389 441437466 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------...
2006 Apr 02
2
Cisco 7960 nat problems.
...0115cd9-d0370002-799a069f-51955597@192.168.1.102 Max-Forwards: 70 CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: <sip:1002@192.168.1.102:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102 s=SIP Call t=0 0 m=audio 25584 RTP/AVP 0 8 18 101 c=IN IP4 192.168.1.102 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a...
2015 Mar 16
1
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
...ip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060> > Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3 > CSeq: 6753 INVITE > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, > UPDATE, PRACK, REGISTER, MESSAGE, REFER > Supported: 100rel, timer, replaces, norefersub > Session-Expires: 1800 > in-SE: 90 > Content-Type: application/sdp > Content-Length: 239 > > v=0 > o=- 1014372762 1014372762 IN IP4 192.168.13.121 > s=Asterisk > c=IN IP4 18.18.19.123 > t=0 0 > m=audio 11614 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:1...
2012 Dec 10
1
Problem with SIP trunk I've set up between two * boxes.
...2.17.0.17> Contact: <sip:6110 at 172.17.9.1:55388;ob> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS CSeq: 24152 INVITE Route: <sip:172.17.0.17;transport=udp;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_d2vzw-16/r1916 Content-Type: application/sdp Content-Length: 354 v=0 o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1 t=0 0 m=audio 4006 RTP/AVP 96 3 0 8 101 c=IN IP4 172.17.9.1 a=rtcp:4007 IN IP4 172.17.9.1 a=sendr...
2016 Dec 16
2
183 Session in Progress. Disconnecting channel for lack of RTP activity
...S5QVC To: sip:8xxx6yyy621 at txxx37.ru Contact: "007" <sip:login at 11.111.11.11:5060;ob> Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg CSeq: 10072 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 90 Min-SE: 90 User-Agent: Keenetic Plus DECT Authorization: Digest username="login", realm="ruvoip.net", nonce="1481885583/bcb53e85a740689479f116a96fc7086b", uri="sip:8xxx6yyy621 at txxx37.ru", response="843f8211896b5b05fcf3a633d6d8eedf&...
2009 Sep 30
0
PBXNSIP Registration Issue
...6b08ea8b68;rport From: "3210" <sip:3210 at 192.168.100.72;user=phone>;tag=63019 To: <sip:6463 at 192.168.100.72;user=phone> Call-ID: 26f9193e at pbx CSeq: 23974 INVITE Max-Forwards: 70 Contact: <sip:TEST1 at 192.168.100.98:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 P-Asserted-Identity: "TEST1" <sip:TEST1 at 192.168.100.72> Content-Type: application/sdp Content-Length: 196 v=0 o=- 30939 30939 IN IP4 192.168...
2015 Mar 15
0
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
...44.194> Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3 CSeq: 6753 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 in-SE: 90 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1014372762 1014372762 IN IP4 192.168.13.121 s=Asterisk c=IN IP4 18.18.19.123 t=0 0 m=audio 11614 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:15...
2017 Jun 05
3
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things
2017 Jun 15
2
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
...1111111111 at sip.easybell.de>;tag=f045584d-da09-4913-9b46-102361e397f2 CSeq: 10 INVITE Call-ID: 7f582402-0ce9-4a1a-87f6-b8de8b2a7bc8 Max-Forwards: 68 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 265 Contact: <sip:64A510CA-5942027B00065C24-6F93C700 at 195.185.37.60;transport=udp> v=0 o=- 1935061780 1935061784 IN IP4 195.185.37.60 s=- c=IN IP4 195.185.37.60 t=0 0 m=image 33818 UDPTL t38 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=...
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
...;transport=UDP> From: "771"<sip:771 at testers.com;transport=UDP>;tag=41030177 Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 AST.ER.ISK.IP s=Z c=IN IP4 AST.ER.ISK.IP t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpma...
2013 Sep 26
0
Asterisk / SIP-Call / AGI-Script / SIGHUP after Answer
...rg>;tag=4313e82f4af9423bab056113e5e05713 To: <sip:3 at myhost.org> Contact: <sip:03794281 at 192.168.1.2:51861> Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink 0.5.0 (Windows) Content-Type: application/sdp Content-Length: 386 v=0 o=- 3589198761 3589198761 IN IP4 192.168.1.2 s=Blink 0.5.0 (Windows) c=IN IP4 192.168.1.2 t=0 0 m=audio 10054 RTP/AVP 108 99 98 9 0 8 96 c=IN IP4 192.168.1.2 a=rtcp:10055 a=rtpmap:108 opus/48000 a=rtpmap:...
2017 Jan 24
2
Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?
...xxx.xxx.xxx>;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M CSeq: 102 INVITE^M Server: Asterisk PBX 14.2.1^M Contact: <sip:xxx.xxx.xxx.xxx:5060>^M Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER^M Supported: 100rel, timer, replaces, norefersub^M Content-Type: application/sdp^M Content-Length: 179^M ^M v=0^M o=- 32730859 32730861 IN IP4 xxx.xxx.xxx.xxx^M s=Asterisk^M c=IN IP4 xxx.xxx.xxx.xxx^M t=0 0^M m=audio 19384 RTP/AVP 0^M a=rtpmap:0 PCMU/8000^M a=ptime:20^M a=maxptime:150^M a=sendrecv^M ACK sip:xxx.xxx.xxx.xxx:5060 SIP/2.0^M Via:...
2018 Oct 03
2
Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"
...XXX.XXX.XXX%20> >;tag=b4134118-08f4-4dbc-a145-573d04438092 CSeq: 2223 INVITE Server: Asterisk PBX 13.20.0 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Contact: <sip:YYY.YYY.YYY.YYY:5060> Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 181 v=0 o=- 11264000 11264002 IN IP4 YYY.YYY.YYY.YYY s=Asterisk c=IN IP4 192.168.11.176 t=0 0 m=audio 18380 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:150 a=sendrecv Receive ACK sip:XXX.XXX.XXX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YY...
2017 Jun 14
2
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++
2015 Mar 15
4
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @