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 147.135.28.128 : 5060 (no NAT)
Using INVITE request as basis request - 110187-11@147.135.28.128
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 147.135.28.250:14704
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 147.135.28.250:14704
Looking for BVLINE in from-broadvoice (domain 192.168.10.192)
<--- Reliably Transmitting (NAT) to 147.135.28.128:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.28.128:5060;received=147.135.28.128
From: "CELLCALLERID"<sip:CELLPHONE@147.135.28.128;user=phone>;tag=nprs
To: "BVCALLERID"<sip:s@EXTERNIP>;tag=as0d3afbf8
Call-ID: 110187-11@147.135.28.128
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 '110187-11@147.135.28.128' in 32000
ms (Method: INVITE)
localhost*CLI>
<--- SIP read from 147.135.28.128:5060 --->
ACK sip:s@EXTERNIP:5060 SIP/2.0
Call-ID: 110187-11@147.135.28.128
CSeq: 1 ACK
From: "CELLCALLERID"<sip:CELLPHONE@147.135.28.128;user=phone>;tag=nprs
To: "BVCALLERID"<sip:s@EXTERNIP>;tag=as0d3afbf8
Via: SIP/2.0/UDP 147.135.28.128:5060;received=EXTERNIP
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '110187-11@147.135.28.128' Method: ACK
[Mar 6 00:44:41] NOTICE[6220]: chan_sip.c:7055 sip_reregister: --
Re-registration for BVLINE@sip.broadvoice.com@sip.broadvoice.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.28.128:5060:
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: 3f30be0279c4e07f0065d5e57a888456@sip.broadvoice.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:s@192.168.10.192>
Event: registration
Content-Length: 0
---
localhost*CLI>
<--- SIP read from 147.135.28.128:5060 --->
SIP/2.0 200 OK
Call-ID: 3f30be0279c4e07f0065d5e57a888456@sip.broadvoice.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:s@192.168.10.192:5060>
Expires: 30
Event: registration
Content-Length: 0
<------------->