Displaying 9 results from an estimated 9 matches for "noresource".
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nodesource
2005 May 10
1
SIP transfers failing
...01618313800@194.24.251.3>;tag=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 SI...
2010 Jan 20
2
Call Xfer issue between DataCenter and User Site
...gt;;tag=as4f7c4d0c
To: <sip:172 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]
<-...
2004 Oct 07
0
Missing Request URI in SIP message
...d" <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=as51f54c64
Call-ID: 11cd8bc246bd1cb0@192.168.2.132
CSeq: 103 BYE
Route: <sip:219@192....
2008 Apr 11
0
problems in REFER request to a different machine
...o: <sip:0778 at 201.73.67.7:5080>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
Contact: <sip:3130296800 at 201.73.67.5>
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=15651
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14
SIP/2.0 200 OK
---
set_destination: Parsing <sip:201.73.67.7:5080> for address/port to
send to
set_destination: set destination to 201.73.67.7, port 5080
Reliably Transmitting (no NAT) to 201.73.67.7:5080:
BYE sip:201.73.67.7:5080...
2011 Mar 04
2
Asterisk <-> Lync / Call Center Transfer / Refer
...:1173 at lyncserver.internal.domain: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 =====================...
2008 Feb 01
2
Asterisk 1.6 - Problems with SIP/REFER
...Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 104 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: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 16
SIP/2.0 200 Ok
---
Scheduling destruction of SIP dialog '7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1' in 32000 ms (Method: REFER)...
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.
2004 Dec 15
1
Help with transferring a second call from a snom 190
...29;user=phone>;tag=as0cd6cdee
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;r...
2004 Dec 15
1
Re: Asterisk-Users Digest, Vol 5, Issue 219
...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):
>>
>...