Generally, what am I looking for when turning SIP debug on? More specifically, the provider says that I'm returning a 404 when they try to call me. Now, I had inbound working, literally, the other day. Outbound works fine. I "may" have broken it either through Asterisk config or the providers portal with settings. Ok, I broke it -- not sure how.
comments interspersed: mordor*CLI> Reliably Transmitting (NAT) to 192.76.120.10:5060: I think/infer/assume that this is the IP address for telnyx SIP servers OPTIONS sip:sip.telnyx.com SIP/2.0 What does OPTIONS mean? Via: SIP/2.0/UDP <externip>:5060;branch=z9hG4bK28142189;rport rport relates to NAT? The message is via SIP UPD from my externip .... what is branch? Max-Forwards: 70 70 hops max? From: "asterisk" <sip:asterisk@<externip>>;tag=as1a7aca46 from my externip, with a hash to keep the calls straight? To: <sip:sip.telnyx.com> easy, to telnyx Contact: <sip:asterisk@<externip>:5060> from me Call-ID: 6fce72627f253b7f2e15dac713b52392@<externip>:5060 another hashcode, Call-ID ? CSeq: 102 OPTIONS ? User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3 easy enough, my system Date: Wed, 06 Jul 2016 02:17:12 GMT easy, date Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE enumerating accepted replies? Supported: replaces ? Content-Length: 0 no data, just "hi" --- mordor*CLI> If I see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions in a SIP trace, that's relatively clear. But what am I looking for with regards to receiving calls? thanks, Thufir
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