search for: touk

Displaying 7 results from an estimated 7 matches for "touk".

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2008 Jan 04
1
Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
...l/poczta at routing-sip' (cause = 66) What could be the reason for this? Thank You in advance bests -tomasz <--- SIP read from 192.168.0.165:7060 ---> SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.0.165:7070;branch=z9hG4bK274b7d50;rport=7070 Contact: <sip:poczta at voip.rd.touk.pl:7170> To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c From: "IPFon"<sip:0225761853 at 192.168.0.165:7070>;tag=as66773d49 Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165 CSeq: 102 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 &l...
2008 Jan 28
0
mwi with sip
...found: <-------------> [Jan 28 11:49:02] --- (19 headers 0 lines) --- [Jan 28 11:49:02] Creating new subscription [Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT) [Jan 28 11:49:02] Found peer 'hellboy' [Jan 28 11:49:02] Looking for hellboy in routing-sip (domain ms.sip.rd.touk.pl) [Jan 28 11:49:02] <--- Transmitting (no NAT) to 192.168.129.38:7060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.129.38:7060;branch=z9hG4bKadcf.1752dae4.0;received= 192.168.129.38 Via: SIP/2.0/UDP 192.168.0.165:7360;rport=7360;branch=z9hG4bKdxcekurc From: "hellboy" <si...
2008 Feb 01
1
meetme music on hold - when only conference member problem
Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten => s,n,MeetMe(|cdIMps) Kind regards tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/a...
2008 Feb 05
0
meetme with ztxen - WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device
...en module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] -- <SIP/sip.rd.touk.pl-b0006fc0> Playing 'conf-invalid' (language 'en') [Feb 5 17:46:17] -- <SIP/sip.rd.touk.pl-b0006fc0> Playing 'conf-getconfno' (language 'en') [Feb 5 17:46:26] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:26]...
2007 Nov 20
1
Realtime - mysql query gives wrong results??
...---+----------+---------+-----------+-----+------+--------+------+-------------+------+---------+-------------+------------+----------------+--------+------+----------+----------+-------------------------+-------------+------------+--------+----------+----------------+--------+ | 3 | outbound-voip.touk.pl | NULL | NULL | NULL | TouK S.K.A | no | NULL | NULL | NULL | NULL | NULL | NULL | 192.168.0.74 | NULL | NULL | NULL | NULL | no | NULL | NULL | NULL | NULL | | NULL | NULL | NULL | NULL |...
2007 Dec 21
1
Asterisk SIP handling - why 491 Request Pending response
...60;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: "IPFon" <sip:0225761853 at 192.168.129.74:5070>;tag=as7217acbc To: <sip:tzl at voip.touk.pl>;tag=as7217acbc Call-ID: 307fda656066a7e264e85cea0742e601 at 192.168.129.74 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16...
2008 Jan 15
0
sip channel error - extension pattern matching problem
Hi, When I have the following extension matching defined: exten => _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --) Asterisk doesn't find it when it receives such SIP request: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:an_hellboy at ms.sip.rd.touk.pl SIP/2.0 Record-Route: <sip:192.168.129.38:7160;lr=on> ... for instance when I use such extension: exten => _vm_.,1,NoOp(-- Context routing-sip-voicemail for ${EXTEN} --) Asterisk finds extensions for RURI like: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:vm_hellboy at...