Here's my issue: I am using Mikrotik for the gateway router with Asterisk behind it. All has been fine until recently as my VoIP provider stopped providing IAX and only does SIP. I changed to SIP and only have 1-way audio. I've been going back and forth verifying my Piaf configuration with my SIP provider and it ended up with the item in question being my router. I am behind a NAT, (as you would guess), and have SIP port forwarding in place but do not have a HUGE range of RTP ports forwarded. Read further to understand why.
Here's my first question: If you look at the attached SIP debug info it seem pretty clear that my SIP messages are getting changed when they reach my SIP provider. I have NO idea how this is happening. My Mikrotik has next to no programming in it for my home office. It simply NATs inside from outside, has IAX and SIP forwarding in place. ??? Has this happened to other people using Mikrotik? Here's my second question: Let's assume I need to replace my Mikrotik. I was wondering what everyone uses for routers. If you would be so kind you can answer this question at http://www.surveymonkey.com/s/WSH6B2D This is simply a survey asking "what router do you use with Asterisk?". I will be happy to share the results. If I forgot your brand, please forgive me and add it in the "other" selection. Thanks to all who reply. NOTE: the SIP message account, provider url and public IP are changed as I did not have their permission to post this. This is what my Voip provider is seeing. Here's why I do not have a TON of RTP ports forwarded. I am glad the provider is willing to deal with this as the RTP range is so wide. They say I should be setting Piaf with no externip= value and let them worry about NAT. Comment from SIP provider: There IS externip or externhost parameter set. Look at registration request from you, it has external IP address in Via and Contact headers: My reply to SIP provider with SIP debug attached. This is what Asterisk says it is sending. This was copied from a SSH session at CLI> REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 38.99.70.232:5060: REGISTER sip:sip.PROVIDER.com SIP/2.0 Via: SIP/2.0/UDP 192.168.111.110:5060;branch=z9hG4bK75f7e979;rport From: <sip:(MY_ACCOUNT)@sip.PROVIDER.com>;tag=as260ed57a To: <sip:[email protected]> Call-ID: [email protected] CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="MY_ACCOUNT", realm="sip.PROVIDER.com", algorithm=MD5, uri="sip:sip.PROVIDER.com", nonce="4b4a910900001d4f8bde7829027049cd6a7e67724f78fe10", response="6c91845c0fb80d385dbcc17a48a0a189" Expires: 120 Contact: <sip:[email protected]> Event: registration Content-Length: 0 Ha, looks like your router knows something about SIP protocol, substitutes addresses and breaks everything :-) Here is what server getting for the same request (call id [email protected]) : Jan 10 21:43:18 t2h /usr/local/sbin/kamailio[19956]: SIP message from udp:72.198.223.192:5060 REGISTER sip:sip.PROVIDER.com SIP/2.0 Via: SIP/2.0/UDP MY_PUBLIC_IP:5060;branch=z9hG4bK75f7e979;rport From: <sip:[email protected]>;tag=as260ed57a To: <sip:[email protected]> Call-ID: [email protected] CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 69 Authorization: Digest username="MY_ACCOUNT", realm="sip.PROVIDER.com", algorithm=MD5, uri="sip:sip.PROVIDER.com", nonce="4b4a 910900001d4f8bde7829027049cd6a7e67724f78fe10", response="6c91845c0fb80d385dbcc17a48a0a189" Expires: 120 Contact: <sip:s...@my_public_ip:5060> Event: registration Content-Length: 0 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
