Displaying 14 results from an estimated 14 matches for "sipfrag".
2008 Feb 01
2
Asterisk 1.6 - Problems with SIP/REFER
...lt;sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at 10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 21
SIP/2.0 183 Ringing
---
-- Stopped music on hold on SIP/5253-0823eab0
set_destination: Parsing <sip:5878 at 10.7.10.51:5060> for address/port to send to
set_desti...
2011 Feb 04
1
standalone NOTIFY message handling for Asterisk
...NOTIFY^M
From: <sip:lab_vms at a.b.c.d:5060>;tag=1426753238-7761-21411^M
To: <sip:+user1 at a.b.c.d:5060>^M
Call-ID: 1231979610-5-21411-CmvtCallId^M
Route: <sip:a.b.c.d:5060;transport=udp;lr>^M
Event: message-summary^M
Accept: application/sdp,application/media_control+xml,message/sipfrag^M
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,INFO,PRACK,UPDATE^M
Contact: <sip:a.b.c.d:5060;transport=udp>^M
Max-Forwards: 70^M
Supported: 100rel,timer,histinfo^M
Subscription-State: active^M
MIME-Version: 1.0^M
Content-Type: application/simple-message-summary^M
Content-Length: 40^M
^M
Messa...
2008 Apr 11
0
problems in REFER request to a different machine
...sip:0778 at 201.73.67.7>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
To: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
Contact: <sip:201.73.67.7:5080>
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
CSeq: 15651 REFER
Event: refer
Expires: 300
Accept: message/sipfrag;version=2.0
Allow-Events: presence, refer
Refer-To: sip:5070 at 201.73.67.7:5070
Referred-By: <sip:0778 at 201.73.67.7>
Content-Length: 0
--- (15 headers 0 lines) ---
Transfer to 5070 in from-sip-external
Transfer from 0778 in from-sip-external
Transmitting (no NAT) to 201.73.67.7:5080:
SI...
2011 Jan 05
1
Blind Transfer not working - 1.4.38
...b2aaaa803fd7e451de826e4 at 87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "<incoming mobile number>" <sip:<incoming mobile number>@x.x.x.x>;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
_______________________________________________________________________________________________________________
But as stated above, extension 504 doesn...
2010 Jan 20
2
Call Xfer issue between DataCenter and User Site
...2 at XXX.XXX.XXX.XXX:10036>;tag=726be2fb618280d0i0
Contact: <sip:176 at YYY.YYY.YYY.YYY>
Call-ID: 718a30a4572984a918b88dc64df642ea at YYY.YYY.YYY.YYY
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.1.1
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 49
SIP/2.0 481 Call leg/transaction does not exist
---
[Jan 20 16:43:38] -- Stopped music on hold on SIP/176-09bf9630
[Jan 20 16:43:38]
<--- SIP read from UDP://XXX.XXX...
2004 Oct 07
0
Missing Request URI in SIP message
....168.2.195:5062;rport;branch=z9hG4bK17f004f7
To: <sip:>
From: "David" <sip:219@192.168.2.195>;tag=as51f54c64
Call-ID: 11cd8bc246bd1cb0@192.168.2.132
CSeq: 102 NOTIFY
Contact: <sip:219@192.168.2.195:5062>
User-Agent: PBX Gateway
Event: refer;id=41590
Content-Type: message/sipfrag; version=2.0
Content-Length: 14
Subscription-state: terminated;reason=noresource
SIP/2.0 200 OK
BYE sip: SIP/2.0
Via: SIP/2.0/UDP 192.168.2.195:5062;branch=z9hG4bK4d88ed51
To: "David" <sip:219@192.168.2.195>;tag=2da39d99e5d753cd
From: <sip:839219@192.168.2.195>;tag=as51f54...
2004 Jun 17
3
SJphone regestration problem - Help!
...les", when I new create a profile, no matter
what Profile type I selected(there three type:Direct SIP Calls, Simple
SIP proxy, Calls through SIP Proxy), in the SIP tab, there are only
four settings I can set. The four settings are
1 Use "application/sip" instead of "message/sipfrag for Notify bodies
2 expose software version
3 Restrict caller identity(support varies for proxies from different
vendors
4 use short headers
I installed the SJphone vision 222b on Linux. Is there something simple
I missed? or am I on the wrong direction? Help would be greatly greatly
apprec...
2008 Apr 13
1
Similar option as promiscredir to use in transfer (REFER)
I made a similar question in a previous thread, but there was no
answer, so I think I was not very clear making the question. What I
need is some configuration that works like "promiscredir=yes" in
sip.conf that enables me to do the same thing with transfer (REFER),
letting me transfer a sip call to a non local sip address.
Thanks in advance,
Thiago
Abra sua conta no Yahoo!
2005 May 10
1
SIP transfers failing
...as0eb5392e
To: <sip:1301@10.0.0.82:5060>;tag=g5VVthPSslPbjLib
Contact: <sip:01618313800@194.24.251.3>
Call-ID: 27584e3a339e535209ea89102043184e@194.24.251.3
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Event: refer;id=1
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14
SIP/2.0 200 OK (no NAT) to 10.0.0.82:5060
set_destination: Parsing <sip:1301@10.0.0.82:5060> for address/port to send to
set_destination: set destination to 10.0.0.82, port 5060
Reliably Transmitting:
BYE sip:1301@10.0.0.82:5060 SIP/2.0
Via: SIP/2.0/UDP 194.24....
2008 Apr 09
6
Jumped from 1.2.7 to 1.4.19, missing CLI colors
Hi,
I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all
the deprecated stuff, but everything seems to be working fine now, except
for a little tiny thing. I lost all color in my CLI, which makes it harder
to debug. Is there something that needs doing? I didn't explicitely disable
colorization from the command line, and I did try using nocolor=no in the
config files.
2006 Jan 09
0
SIP-SIP transfer via the REFER/NOTIFY method
...9hG4bKC77
From: ?us_pstn_caller? <sip:63.82.77.14>;tag=2112301987
To: <sip:pstn_6523567@inbound4.vonage.net>;tag=3655015093
Call-ID: CF6FC3F9-800111DA-999DDF96-A870513F@63.82.77.14
CSeq: 102 NOTIFY
Contact: <sip:63.82.77.14:5060>
Max-Forwards: 13
Event: refer
Content-Type: message/sipfrag
Content-Length: 18
SEND TIME: 26288881
SEND >> sphone.vopr.vonage.net:5061
SIP/2.0 200 Ok
Via: SIP/2.0/UDP sphone.vopr.vonage.net:5061
Via: SIP/2.0/UDP inbound4.vonage.net:5060
Via: SIP/2.0/UDP 63.82.77.14:5060;branch=z9hG4bKC77
From: ?us_pstn_caller? <sip:63.82.77.14>;tag=21123019...
2011 Mar 04
2
Asterisk <-> Lync / Call Center Transfer / Refer
...in:5067>;epid=431D53633D;tag=42b6d8c72b
Contact: <sip:500 at 10.10.10.10:5067;transport=TLS>
Call-ID: 4ad0626f79c7c8de66b668b13d624129 at 10.10.10.10:5067
CSeq: 103 NOTIFY
User-Agent: FPBX-2.8.1(1.8.3)
Event: refer;id=2
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 49
SIP/2.0 481 Call leg/transaction does not exist
---
================================= End Asterisk Debug ============================================
2004 Dec 15
1
Help with transferring a second call from a snom 190
...To: "snom_01" <sip:snom_01@192.168.0.129>;tag=jdx5841oim
Contact: <sip:7635551212@192.168.0.129>
Call-ID: 3c30fbaf1388-i20e2ommrm65@192-168-102-70
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14
SIP/2.0 200 OK
------------------------------------------------------------------------
Sent to udp:192.168.0.129:5060 at 14/12/2004 18:22:12:450 (358 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.129:5060;branch=z9hG4bK2214bb84;rport=5060
From: <sip:763555...
2004 Dec 15
1
Re: Asterisk-Users Digest, Vol 5, Issue 219
...;;tag=jdx5841oim
>>Contact: <sip:7635551212@192.168.0.129>
>>Call-ID: 3c30fbaf1388-i20e2ommrm65@192-168-102-70
>>CSeq: 102 NOTIFY
>>User-Agent: Asterisk PBX
>>Event: refer;id=4
>>Subscription-state: terminated;reason=noresource
>>Content-Type: message/sipfrag;version=2.0
>>Content-Length: 14
>>
>>SIP/2.0 200 OK
>>
>>------------------------------------------------------------------------
>>
>>Sent to udp:192.168.0.129:5060 at 14/12/2004 18:22:12:450 (358 bytes):
>>
>>SIP/2.0 200 Ok
>>Via: SIP/2...