I have debugging on in Asterisk and "sip debug". How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric
It's the "From: " line. Mark On Thu, 6 Mar 2003, Eric Wieling wrote:> I have debugging on in Asterisk and "sip debug". > > How do I tell what username a SIP client is trying to use to > register with Asterisk as? > > --Eric > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
How i must setup h323 on asterisk? Juan -----Mensaje original----- De: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com]En nombre de Eric Wieling Enviado el: Jueves, 06 de Marzo de 2003 03:56 p.m. Para: asterisk-users at lists.digium.com Asunto: [Asterisk-Users] SIP Debugging I have debugging on in Asterisk and "sip debug". How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric _______________________________________________ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Hello, When I enable SIP debugging I receive: Peer RTP is at port 10.10.60.16:0 Shouldn't the RTP port be a number between 10000 - 20000? - Brent
Hi! I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. There is no log message, why this packet is not accepted/processed. Is there a ways to increase the sip debugging? thanks klaus Retransmitting #5 (NAT) to 192.174.68.4:5060: CANCEL sip:431505641636@192.174.68.4 SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: "klaus1" <sip:00437248600777@213.174.230.213>;tag=as4233f839 To: <sip:431505641636@192.174.68.4> Contact: <sip:00437248600777@213.174.230.213> Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f@213.174.230.213 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI> <-- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: "klaus1" <sip:00437248600777@213.174.230.213>;tag=as4233f839 To: <sip:431505641636@192.174.68.4>;tag=2870350146 Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f@213.174.230.213 CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] --- (7 headers 0 lines)--- Destroying call '4c614a3a4a2d5ba94ea4be5e62c3d37f@213.174.230.213' Retransmitting #6 (NAT) to 192.174.68.4:5060: CANCEL sip:431505641636@192.174.68.4 SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: "klaus1" <sip:00437248600777@213.174.230.213>;tag=as4233f839 To: <sip:431505641636@192.174.68.4> Contact: <sip:00437248600777@213.174.230.213> Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f@213.174.230.213 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI> <-- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: "klaus1" <sip:00437248600777@213.174.230.213>;tag=as4233f839 To: <sip:431505641636@192.174.68.4>;tag=2870350146 Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f@213.174.230.213 CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123]