search for: 524302

...was gone, without me doing anything..Has anyone observed this thing before... Called 810 -- SIP/810-b6dc is ringing -- SIP/810-b6dc answered SIP/910-6c4e -- Attempting native bridge of SIP/910-6c4e and SIP/810-b6dc WARNING[1227879616]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 = 524302 is not codec1 = 524302, can't do reinvite -- Got SIP response 482 "Loop Detected" back from 129.82.44.226 WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 3c2706bad222-n2s56u19hj1l@129-82-44-226 for seqno 1 (Response) WARNING[114210656...

...AW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 269/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.21 From: "BC IP Phone" <sip:btel@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079 To: <si...

...3.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include pr...

...udio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd...

...s this normal? Why just one phone (a Grandstream Handytone ATA)? Running "sip show channel 984ee48048d" I get the output below so it seems "active": * SIP Call Direction: Incoming Call-ID: 984ee48048d9e01c@192.168.0.22 Our Codec Capability: 524302 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format UNKN Theoretical Address: 192.168.0.22:5060 Received Address: 192.168.0.22:5060 NAT Support: RFC3581 Our Tag: 1190462248 Their Tag:...

Hello I have a problem, I installed Mandriva 2008.1 spring edition and installed wine 0.9.60 and cedega 6.0.5. I have no sound under wine with ALSA and I have sound under OSS. BUT all work with cedega My game "RUNAWAY" say something like directx can't use sound beause it is already by another app. I Tryed some ticks I found under some forums but nothing work , ALSACONF say 2

...audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 524302, them - 1550/0, combined - 14 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A From: Tony Hoyle <sip:6001@asterisk>;tag=375...

I am trying to figure out the correct syntax for the CLI command "SIP SHOW CHANNELS". I have tried SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such as: -- Zap/15-1 is ringing -- Zap/15-1 answered SIP/206-4299 asterisk*CLI> sip show channel SIP/206-4299 No such SIP Call ID 'SIP/206-4299' I always get the "No such SIP

...est as basis request Sending to 213.137.73.176 : 5060 (non-NAT) Found audio format 4 Found audio format 18 Found audio format 101 Found audio format 19 Found description format G723 Found description format G729 Found description format telephone-event Found description format CN Capabilities: us - 524302, them - 257/0, combined - 0 Non-codec capabilities: us - 1, them - 3, combined - 1 Sip read: INVITE sip:14103445557@162.33.165.198 SIP/2.0 Record-Route: <sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176> Via: SIP/2.0/UDP 213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-...

...a71b2@192.168.1.190 CSeq: 53320 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 266 [ ... ] Sending to 192.168.1.190 : 5060 (non-NAT) [ ... ] Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 DEBUG[5126]: File chan_sip.c, Line 3965 (check_user): Setting NAT on RTP to 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required [ ... ] ACK sip:2@192.168.1.10 SIP/2.0 [ ... ] DEBUG[5126]:...

...12 s=Sip Call t=0 0 m=audio 10014 RTP/AVP 0 101 c=IN IP4 62.254.245.12 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16 14 headers, 9 lines Found audio format 0 Found audio format 101 Found description format PCMU Found description format telephone-event Capabilities: us - 524302, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: <sip:62.254.245.14:5060;lr=1> list_route: hop: <sip:3046@vega.culver-tec.com:5060;maddr=62.254.245.12> set_destination: Parsing <sip:62.254.245.14:5060;lr=1> for address/port to send...

...000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 12 headers, 13 lines Using latest request as basis request Sending to 10.0.1.202 : 5060 (non-NAT) Found audio format UNKN <cut> Found description format PCMU <cut> Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 5703 in johnhome list_route: hop: <sip:5702@10.0.1.202;user=phone> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.202;received=10.0.1.202 +++++++++++++++++++++++++ divider inserted...