rendered paste bodyTrying internal call 100 -> 101
Asterix+FreePBX (192.168.1.2)
100: Cisco 7941 (192.168.1.158)
101: Grandstream HT486 (192.168.1.3)
Call is dropped with busy signal due to "603 Declined", see debug below.
---8<---
<--- SIP read from 192.168.1.158:49182 --->
INVITE sip:101@miraculix.int.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbbb04ded
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
Max-Forwards: 70
Date: Wed, 30 Jan 2008 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7941G/8.0
Contact: <sip:100@192.168.1.158:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21243 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 30194 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 13 lines) ---
Sending to 192.168.1.158 : 5060 (no NAT)
Using INVITE request as basis request - 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
<--- Reliably Transmitting (no NAT) to 192.168.1.158:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbbb04ded;received=192.168.1.158
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>;tag=as6fc5b9bc
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7736ab5a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158' in 32000 ms (Method: INVITE)
Found user '100'
miraculix*CLI>
<--- SIP read from 192.168.1.158:50468 --->
ACK sip:101@miraculix.int.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbbb04ded
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>;tag=as6fc5b9bc
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
Date: Wed, 30 Jan 2008 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
miraculix*CLI>
<--- SIP read from 192.168.1.158:49182 --->
INVITE sip:101@miraculix.int.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
Max-Forwards: 70
Date: Wed, 30 Jan 2008 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7941G/8.0
Contact: <sip:100@192.168.1.158:5060;transport=udp>
Proxy-Authorization: Digest username="100",realm="asterisk",uri="sip:101@miraculix.int.mydomain.com",response="9049ba2387fb625bb0d087c32128d18a",nonce="7736ab5a",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21243 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 30194 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 13 lines) ---
Sending to 192.168.1.158 : 5060 (no NAT)
Using INVITE request as basis request - 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
Found user '100'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.158:30194
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.158:30194
Looking for 101 in from-internal (domain miraculix.int.mydomain.com)
list_route: hop: <sip:100@192.168.1.158:5060;transport=udp>
miraculix*CLI>
<--- Transmitting (no NAT) to 192.168.1.158:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a;received=192.168.1.158
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.1.2>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 192.168.1.158:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a;received=192.168.1.158
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>;tag=as1f7d6359
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.1.2>
Content-Length: 0
<------------>
miraculix*CLI>
<--- SIP read from 192.168.1.158:50469 --->
ACK sip:101@miraculix.int.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a
From: "100" <sip:100@miraculix.int.mydomain.com>;tag=001aa17bdcb002914ca32cfc-074f6117
To: <sip:101@miraculix.int.mydomain.com>;tag=as1f7d6359
Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158
Date: Wed, 30 Jan 2008 GMT
CSeq: 102 ACK
Content-Length: 0
---8<---
Relevant snippet from extensions_additional.conf:
---8<---
[ext-local]
include => ext-local-custom
exten => 100,1,Macro(exten-vm,100,100)
exten => 100,n,Hangup
exten => 100,hint,SIP/100
exten => ${VM_PREFIX}100,1,Macro(vm,100,DIRECTDIAL)
exten => ${VM_PREFIX}100,n,Hangup
exten => vmb100,1,Macro(vm,100,BUSY)
exten => vmb100,n,Hangup
exten => vmu100,1,Macro(vm,100,NOANSWER)
exten => vmu100,n,Hangup
exten => vms100,1,Macro(vm,100,NOMESSAGE)
exten => vms100,n,Hangup
exten => 101,1,Macro(exten-vm,101,101)
exten => 101,n,Hangup
exten => 101,hint,SIP/101
exten => ${VM_PREFIX}101,1,Macro(vm,101,DIRECTDIAL)
exten => ${VM_PREFIX}101,n,Hangup
exten => vmb101,1,Macro(vm,101,BUSY)
exten => vmb101,n,Hangup
exten => vmu101,1,Macro(vm,101,NOANSWER)
exten => vmu101,n,Hangup
exten => vms101,1,Macro(vm,101,NOMESSAGE)
exten => vms101,n,Hangup
---8<---