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 @