Well I’ve successfully managed to make 2 polycom 501’s work internally VM/Extensions. I’ve also got outbound calls working with broadvoice but for the life of me can not figure out what I’m doing wrong with inbound calls. My suspicion would be that I’m doing something wrong with NAT but I’ve tried both externip / localnet which for some reason appear to be mutually exclusive (1 or the other). The asterisk box is in a DMZ behind a single wireless router. For brevity I’ve included only what I thought was pertinent. As I understand it the end of the register statement is the local extension the call should match. I’ve tried both the BV phone number, s, and 101 (the first extension in the dial plan). Also under the sip.broadvoice.com is the context from-broadvoice to which I understand those calls are “routed”. Right now when I call the BV number it acts like BV can’t route the call or maybe its not matching any dial plan entry so it says something like “The caller you are trying to reach cannot be reached at this time.” If anyone can lend a hand I would greatly appreciate it. Thanks!
– SIP.CONF –
[general] srvlookup=yes pedantic=no context=default callerid=Unknown externip=MYIP ;localnet=192.168.10.0/255.255.255.0 register => PHONE@sip.broadvoice.com:PASS:PHONE@sip.broadvoice.com/PHONE [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PHONE secret=PASS username=PHONE insecure=very context=from-broadvoice authname=PHONE dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no callerid=PHONE nat=yes
– EXTENSIONS.CONF –
[from-broadvoice] exten => s,1,Answer() exten => s,2,Playback(hello-world) exten => s,3,Hangup()
--- (10 headers 14 lines) --- Sending to 184.108.40.206 : 5060 (no NAT) Using INVITE request as basis request - email@example.com Found peer 'sip.broadvoice.com' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Peer audio RTP is at port 220.127.116.11:14704 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G726-32 for ID 2 Found description format G729 for ID 18 Found description format iLBC for ID 96 Found description format t38 for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd0c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 18.104.22.168:14704 Looking for BVLINE in from-broadvoice (domain 192.168.10.192) <--- Reliably Transmitting (NAT) to 22.214.171.124:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 126.96.36.199:5060;received=188.8.131.52 From: "CELLCALLERID"<sip:CELLPHONE@184.108.40.206;user=phone>;tag=nprs To: "BVCALLERID"<sip:s@EXTERNIP>;tag=as0d3afbf8 Call-ID: firstname.lastname@example.org CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'email@example.com' in 32000 ms (Method: INVITE) localhost*CLI> <--- SIP read from 220.127.116.11:5060 ---> ACK sip:s@EXTERNIP:5060 SIP/2.0 Call-ID: firstname.lastname@example.org CSeq: 1 ACK From: "CELLCALLERID"<sip:CELLPHONE@18.104.22.168;user=phone>;tag=nprs To: "BVCALLERID"<sip:s@EXTERNIP>;tag=as0d3afbf8 Via: SIP/2.0/UDP 22.214.171.124:5060;received=EXTERNIP Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog 'email@example.com' Method: ACK [Mar 6 00:44:41] NOTICE: chan_sip.c:7055 sip_reregister: -- Re-registration for BVLINE@firstname.lastname@example.org REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 126.96.36.199:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.192:5060;branch=z9hG4bK3d9318e8;rport From: <sip:BVLINE@sip.broadvoice.com>;tag=as39b3d8a7 To: <sip:BVLINE@sip.broadvoice.com> Call-ID: email@example.com CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:firstname.lastname@example.org> Event: registration Content-Length: 0 --- localhost*CLI> <--- SIP read from 188.8.131.52:5060 ---> SIP/2.0 200 OK Call-ID: email@example.com CSeq: 106 REGISTER From: <sip:BVLINE@sip.broadvoice.com>;tag=as39b3d8a7 To: <sip:BVLINE@sip.broadvoice.com> Via: SIP/2.0/UDP 192.168.10.192:5060;branch=z9hG4bK3d9318e8 Contact: <sip:firstname.lastname@example.org:5060> Expires: 30 Event: registration Content-Length: 0 <------------->