Marko Rakar
2004-Mar-25 06:22 UTC
[Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device
I have tried to connect asterisk (which I use through hisax isdn4linux device) with mediatrix sip device with g729 codec asterisk can not connect with mediatrix (it connects when ulaw/alaw are used) when g729 is forced any ides what to do? Sip read: SIP/2.0 200 OK Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 102 INVITE From: 0 <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 178 Content-Type: application/sdp Contact: <sip:301@192.168.3.211> Allow: INVITE, ACK, BYE, CANCEL, REFER v=0 o=MxSIP 0 0 IN IP4 192.168.3.211 s=SIP Call c=IN IP4 192.168.3.211 t=0 0 m=audio 5004 RTP/AVP 18 8 0 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 10 headers, 9 lines Found audio format UNKN Found audio format ALAW Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Capabilities: us - 268, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: <sip:301@192.168.3.211> set_destination: Parsing <sip:301@192.168.3.211> for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Transmitting: ACK sip:301@192.168.3.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: "0" <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Contact: <sip:0@192.168.3.6> Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 Mar 25 15:24:26 NOTICE[1225991360]: channel.c:1513 ast_set_write_format: Unable to find a path from UNKN to SLINR Mar 25 15:24:26 WARNING[1225991360]: channel.c:1920 ast_channel_make_compatible: No path to translate from Modem[i4l]/ttyI0(64) to SIP/301-3309(256) Mar 25 15:24:26 WARNING[1225991360]: app_dial.c:702 dial_exec: Had to drop call because I couldn't make Modem[i4l]/ttyI0 compatible with SIP/301-3309 set_destination: Parsing <sip:301@192.168.3.211> for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Reliably Transmitting: BYE sip:301@192.168.3.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: "0" <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Contact: <sip:0@192.168.3.6> Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 103 BYE From: 0 <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 0 7 headers, 0 lines ---- Sometimes you're the bug, sometimes you're the windshield. mailto:marko@printel.hr http://printel.hr
Wes Marderness
2004-Mar-25 07:46 UTC
[Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device
You need a G729 license for asterisk to make a connection. You have to get them from diguim, they are $10 a channel. They do give you a single channel demo license, you just have to get it from them. Wes -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Marko Rakar Sent: Thursday, March 25, 2004 8:23 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device I have tried to connect asterisk (which I use through hisax isdn4linux device) with mediatrix sip device with g729 codec asterisk can not connect with mediatrix (it connects when ulaw/alaw are used) when g729 is forced any ides what to do? Sip read: SIP/2.0 200 OK Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 102 INVITE From: 0 <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 178 Content-Type: application/sdp Contact: <sip:301@192.168.3.211> Allow: INVITE, ACK, BYE, CANCEL, REFER v=0 o=MxSIP 0 0 IN IP4 192.168.3.211 s=SIP Call c=IN IP4 192.168.3.211 t=0 0 m=audio 5004 RTP/AVP 18 8 0 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 10 headers, 9 lines Found audio format UNKN Found audio format ALAW Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Capabilities: us - 268, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: <sip:301@192.168.3.211> set_destination: Parsing <sip:301@192.168.3.211> for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Transmitting: ACK sip:301@192.168.3.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: "0" <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Contact: <sip:0@192.168.3.6> Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 Mar 25 15:24:26 NOTICE[1225991360]: channel.c:1513 ast_set_write_format: Unable to find a path from UNKN to SLINR Mar 25 15:24:26 WARNING[1225991360]: channel.c:1920 ast_channel_make_compatible: No path to translate from Modem[i4l]/ttyI0(64) to SIP/301-3309(256) Mar 25 15:24:26 WARNING[1225991360]: app_dial.c:702 dial_exec: Had to drop call because I couldn't make Modem[i4l]/ttyI0 compatible with SIP/301-3309 set_destination: Parsing <sip:301@192.168.3.211> for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Reliably Transmitting: BYE sip:301@192.168.3.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: "0" <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Contact: <sip:0@192.168.3.6> Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Call-ID: 1714ebf049da1da918d54b84725aeedb@192.168.3.6 CSeq: 103 BYE From: 0 <sip:0@192.168.3.6>;tag=as01323dfd To: <sip:301@192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 0 7 headers, 0 lines ---- Sometimes you're the bug, sometimes you're the windshield. mailto:marko@printel.hr http://printel.hr _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users