I have asterisk with following users; a) zaphfc ISDN card with two channels b) two mediatrix FXS gateways with four channels each c) 1x CISCO 7905G d) two notebooks with MS Messenger 4.7
Now, it seems that any combination works correctly in all combinations except when I call from MS messenger and then call is dropped always in 25th second of the call. Any ideas what I did wrong? here is my messenger sip.conf portion; [marko] type=friend reinvite=no username=marko host=dynamic mailbox=1300 here is my cisco 7905 sip.conf portion; [123] type=peer reinvite=no callerid= "Marko Rakar" username=123 secret=1234 dtmfmode=inband careinvite=yes host=dynamic defaultip=192.168.3.52 incominglimit=2 outgoinglimit=2 here is a part of my sip debug file 9 headers, 0 lines Sending to 192.168.3.54 : 14250 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.54:14250 From: "marko" <sip:[EMAIL PROTECTED]>;tag=124ece30-95ed-4a45-8a62-5bacd517e1ae To: <sip:[EMAIL PROTECTED];user=phone>;tag=as05865310 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.3.54:14250 set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> for address/port to send to set_destination: set destination to 192.168.3.52, port 5060 We're at 192.168.3.6 port 29312 Answering with preferred capability 4 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235 From: "marko" <sip:[EMAIL PROTECTED]>;tag=as33a10ddc To: <sip:[EMAIL PROTECTED]>;tag=1930002232 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 212 v=0 o=root 5963 5965 IN IP4 192.168.3.6 s=session c=IN IP4 192.168.3.6 t=0 0 m=audio 29312 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.3.52:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235 From: "marko" <sip:[EMAIL PROTECTED]>;tag=as33a10ddc To: <sip:[EMAIL PROTECTED]>;tag=1930002232 Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE Contact: 123 <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Server: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 202 Content-Type: application/sdp v=0 o=123 37649 37649 IN IP4 192.168.3.52 s=Cisco 7905 SIP Call c=IN IP4 192.168.3.52 t=0 0 m=audio 16384 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 9 lines Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format telephone-event Capabilities: us - 12, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> for address/port to send to set_destination: set destination to 192.168.3.52, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235 From: "marko" <sip:[EMAIL PROTECTED]>;tag=as33a10ddc To: <sip:[EMAIL PROTECTED]>;tag=1930002232 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.52:5060 set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> for address/port to send to set_destination: set destination to 192.168.3.52, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235 From: "marko" <sip:[EMAIL PROTECTED]>;tag=as33a10ddc To: <sip:[EMAIL PROTECTED]>;tag=1930002232 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 105 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.52:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK6137f235 From: "marko" <sip:[EMAIL PROTECTED]>;tag=as33a10ddc To: <sip:[EMAIL PROTECTED]>;tag=1930002232 Call-ID: [EMAIL PROTECTED] CSeq: 105 BYE Server: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 9 headers, 0 lines ---- Give someone a fish, you feed him for one day. Teach him how to fish, and you lose a steady customer. mailto:[EMAIL PROTECTED] http://printel.hr _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
