extensions.conf: --------------- [outbound0] exten => 123/4166666666,1,Dial(SIP/foo) [authority0] include => outbound0 sip.conf: -------- [budge2] secret=budge2 type=friend host=dynamic ; This device needs to register canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=ulaw context=authority0 callerid=left <4166666666> mailbox=123@default [foo] secret=wordpass type=friend host=dynamic ; This device needs to register canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=ulaw context=authority0 callerid=right <4165551212> mailbox=456@default ------------------- *CLI> show dialplan [ Context 'authority0' created by 'pbx_config' ] Include => 'outbound0' [pbx_config] [ Context 'outbound0' created by 'pbx_config' ] '123' => 1. Dial(SIP/foo) [pbx_config] [ Context 'parkedcalls' created by 'res_features' ] '700' => 1. Park() [res_features] -= 2 extensions (2 priorities) in 3 contexts. =- *CLI> ------------------- <-- SIP read from 192.168.20.85:5060: INVITE sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK6e6a809896c9ae48 From: ;tag=04eeeadf05ee15bf To: Contact: Supported: replaces Call-ID: f2cd2f1e4c08590c@192.168.20.85 CSeq: 49500 INVITE User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 386 v=0 o=budge2 8000 8000 IN IP4 192.168.20.85 s=SIP Call c=IN IP4 192.168.20.85 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 18 lines) --- Using INVITE request as basis request - f2cd2f1e4c08590c@192.168.20.85 Sending to 192.168.20.85 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.20.85:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK6e6a809896c9ae48;received=192.168.20.85 From: ;tag=04eeeadf05ee15bf To: ;tag=as34e638d8 Call-ID: f2cd2f1e4c08590c@192.168.20.85 CSeq: 49500 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ba66792" Content-Length: 0 --- Scheduling destruction of call 'f2cd2f1e4c08590c@192.168.20.85' in 15000 ms Found user 'budge2' <-- SIP read from 192.168.20.85:5060: ACK sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK6e6a809896c9ae48 From: ;tag=04eeeadf05ee15bf To: ;tag=as34e638d8 Contact: Call-ID: f2cd2f1e4c08590c@192.168.20.85 CSeq: 49500 ACK User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (11 headers 0 lines) --- <-- SIP read from 192.168.20.85:5060: INVITE sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK39ba3cdb185d82db From: ;tag=04eeeadf05ee15bf To: Contact: Supported: replaces Proxy-Authorization: Digest username="budge2", realm="asterisk", algorithm=MD5, uri="sip:123@192.168.20.86:5060", nonce="6ba66792", response="7ccce7a1adaf7548892c16d4cd7fcf76" Call-ID: f2cd2f1e4c08590c@192.168.20.85 CSeq: 49501 INVITE User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 386 v=0 o=budge2 8000 8001 IN IP4 192.168.20.85 s=SIP Call c=IN IP4 192.168.20.85 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (14 headers 18 lines) --- Using INVITE request as basis request - f2cd2f1e4c08590c@192.168.20.85 Sending to 192.168.20.85 : 5060 (non-NAT) Found user 'budge2' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 97 Found RTP audio format 9 Found RTP audio format 101 Peer audio RTP is at port 192.168.20.85:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format iLBC Found description format G722 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 123 in authority0 (domain 192.168.20.86) Reliably Transmitting (no NAT) to 192.168.20.85:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK39ba3cdb185d82db;received=192.168.20.85 From: ;tag=04eeeadf05ee15bf To: ;tag=as34e638d8 Call-ID: f2cd2f1e4c08590c@192.168.20.85 CSeq: 49501 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.20.85:5060: ACK sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK39ba3cdb185d82db From: ;tag=04eeeadf05ee15bf To: ;tag=as34e638d8 Contact: Proxy-Authorization: Digest username="budge2", realm="asterisk", algorithm=MD5, uri="sip:123@192.168.20.86:5060", nonce="6ba66792", response="98fccf4a8e6a6cf189ca5854fac034cc" Call-ID: f2cd2f1e4c08590c@192.168.20.85 CSeq: 49501 ACK User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (12 headers 0 lines) --- Destroying call 'f2cd2f1e4c08590c@192.168.20.85'