Hello everyone,
I need help with freePBX setup.
I have pstn sip trunk from telco, incoming calls are received and i can see them when using "sip set debug on"
I have another sip trunk that is connected to avaya session manager. when debug is on i can see that freepbx is routing calls to avaya session manager.
What i want is to route all calls from pstn sip trunk to avaya session manager trunk. I have made Inbound route that routes all incoming traffic from pstn sip trunk to session manager trunk.
The problem i have is that when the call is routed to avaya session manager trunk To field is changed and there is no called number id only calling number id.
I need called number id so i can make apropriate rules in avaya session manager.
Verison of software is:
Asterisk 11.2.1
FreePBX 2.11.0.0beta2.6
Avaya session manager 6.3
Here is log from sip set debug on.
<— SIP read from UDP:10.0.0.2:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bKrq673s0068mhulk615k0.1
To: "7155301 7155301"sip:[email protected];cscf
From: sip:[email protected];user=phone;tag=619390923-1375264232667-
Call-ID: [email protected]
CSeq: 437348974 INVITE
Max-Forwards: 27
Content-Length: 236
Contact: sip:[email protected]:5060;transport=udp
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY
Accept: multipart/mixed
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-hotsip-FileTransfer+xml
Supported: timer
P-Asserted-Identity: sip:112017088;[email protected];user=phone
Privacy: none
P-Charging-Vector: icid-value=b07d234105d76bf90ad408bb06ee57e
Min-SE: 180
Session-Expires: 1800
P-Called-Party-ID: sip:[email protected]
v=0
o=BroadWorks 90037223 1 IN IP4 10.0.0.2
s=-
c=IN IP4 10.0.0.2
t=0 0
a=sendrecv
m=audio 37218 RTP/AVP 8 18 98
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-15
<------------->
— (22 headers 12 lines) —
Sending to 10.0.0.2:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘tk1’ for ‘2017088’ from 10.0.0.2:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 98
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 98
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.2:37218
Looking for s in from-trunk-sip-tk1 (domain 10.2.4.90)
list_route: hop: sip:[email protected]:5060;transport=udp
<— Transmitting (no NAT) to 10.0.0.2:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bKrq673s0068mhulk615k0.1;received=10.0.0.2
From: sip:[email protected];user=phone;tag=619390923-1375264232667-
To: "7155301 7155301"sip:[email protected];cscf
Call-ID: [email protected]
CSeq: 437348974 INVITE
Server: FPBX-2.11.0beta3(11.2.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
– Executing [s@from-trunk-sip-tk1:1] Set(“SIP/tk1-000000c7”, “GROUP()=OUT_1”) in new stack
– Executing [s@from-trunk-sip-tk1:2] Goto(“SIP/tk1-000000c7”, “from-trunk,s,1”) in new stack
– Goto (from-trunk,s,1)
– Executing [s@from-trunk:1] ExecIf(“SIP/tk1-000000c7”, “1?Set(__FROM_DID=s)”) in new stack
– Executing [s@from-trunk:2] Gosub(“SIP/tk1-000000c7”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/tk1-000000c7”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/tk1-000000c7”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/tk1-000000c7”, “”) in new stack
– Executing [s@from-trunk:3] Set(“SIP/tk1-000000c7”, “CDR(did)=s”) in new stack
– Executing [s@from-trunk:4] ExecIf(“SIP/tk1-000000c7”, “1 ?Set(CALLERID(name)=2017088)”) in new stack
– Executing [s@from-trunk:5] Set(“SIP/tk1-000000c7”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [s@from-trunk:6] Set(“SIP/tk1-000000c7”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [s@from-trunk:7] Goto(“SIP/tk1-000000c7”, “ext-trunk,2,1”) in new stack
– Goto (ext-trunk,2,1)
– Executing [2@ext-trunk:1] Set(“SIP/tk1-000000c7”, “TDIAL_STRING=SIP/Avaya_SMGR”) in new stack
– Executing [2@ext-trunk:2] Set(“SIP/tk1-000000c7”, “DIAL_TRUNK=2”) in new stack
– Executing [2@ext-trunk:3] Goto(“SIP/tk1-000000c7”, “ext-trunk,tdial,1”) in new stack
– Goto (ext-trunk,tdial,1)
– Executing [tdial@ext-trunk:1] Set(“SIP/tk1-000000c7”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [tdial@ext-trunk:2] GotoIf(“SIP/tk1-000000c7”, “1?nomax”) in new stack
– Goto (ext-trunk,tdial,4)
– Executing [tdial@ext-trunk:4] ExecIf(“SIP/tk1-000000c7”, “1?Set(CALLERPRES()=allowed_not_screened)”) in new stack
– Executing [tdial@ext-trunk:5] Set(“SIP/tk1-000000c7”, “DIAL_NUMBER=s”) in new stack
– Executing [tdial@ext-trunk:6] GosubIf(“SIP/tk1-000000c7”, “0?sub-flp-2,s,1()”) in new stack
– Executing [tdial@ext-trunk:7] Set(“SIP/tk1-000000c7”, “OUTNUM=s”) in new stack
– Executing [tdial@ext-trunk:8] Set(“SIP/tk1-000000c7”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [tdial@ext-trunk:9] Dial(“SIP/tk1-000000c7”, “SIP/Avaya_SMGR/s,300,Tt”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 14884
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.2.18.61:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.2.4.90:5060;branch=z9hG4bK7acd437b
Max-Forwards: 70
From: “2017088” sip:[email protected];tag=as1646631c
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060;transport=TCP
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0beta3(11.2.1)
Date: Wed, 31 Jul 2013 09:50:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 34666148 34666148 IN IP4 10.2.4.90
s=Asterisk PBX 11.2.1
c=IN IP4 10.2.4.90
t=0 0
m=audio 14884 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-- Called SIP/Avaya_SMGR/s
<— SIP read from TCP:10.2.18.61:5060 —>
SIP/2.0 100 Trying
Call-ID: [email protected]
CSeq: 102 INVITE
From: “2017088” sip:[email protected];tag=as1646631c
To: sip:[email protected]:5060
Via: SIP/2.0/TCP 10.2.4.90:5060;branch=z9hG4bK7acd437b
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from TCP:10.2.18.61:5060 —>
SIP/2.0 404 Not Found (cannot determine routing domain, check SM ports)
Call-ID: [email protected]
CSeq: 102 INVITE
From: “2017088” sip:[email protected];tag=as1646631c
To: sip:[email protected]:5060;tag=24017248102429256_local.1374087054677_539644_539643
Via: SIP/2.0/TCP 10.2.4.90:5060;branch=z9hG4bK7acd437b
Record-Route: sip:[email protected];lr;transport=TCP
Av-Global-Session-ID: a420a5a0-f9c6-11e2-b88e-005056bc3f1a
Server: AVAYA-SM-6.3.2.0.632023
Contact: sip:10.2.18.63:15060;transport=tcp
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Transmitting (no NAT) to 10.2.18.61:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.2.4.90:5060;branch=z9hG4bK7acd437b
Max-Forwards: 70
From: “2017088” sip:[email protected];tag=as1646631c
To: sip:[email protected]:5060;tag=24017248102429256_local.1374087054677_539644_539643
Contact: sip:[email protected]:5060;transport=TCP
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-2.11.0beta3(11.2.1)
Content-Length: 0
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [tdial@ext-trunk:10] Set(“SIP/tk1-000000c7”, “CALLERID(number)=2017088”) in new stack
– Executing [tdial@ext-trunk:11] Set(“SIP/tk1-000000c7”, “CALLERID(name)=2017088”) in new stack
– Executing [tdial@ext-trunk:12] Hangup(“SIP/tk1-000000c7”, “”) in new stack
== Spawn extension (ext-trunk, tdial, 12) exited non-zero on 'SIP/tk1-000000c7’
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
<— Reliably Transmitting (no NAT) to 10.0.0.2:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bKrq673s0068mhulk615k0.1;received=10.0.0.2
From: sip:[email protected];user=phone;tag=619390923-1375264232667-
To: "7155301 7155301"sip:[email protected];cscf;tag=as5554baa3
Call-ID: [email protected]
CSeq: 437348974 INVITE
Server: FPBX-2.11.0beta3(11.2.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<— SIP read from UDP:10.0.0.2:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bKrq673s0068mhulk615k0.1
CSeq: 437348974 ACK
To: "7155301 7155301"sip:[email protected];cscf;tag=as5554baa3
From: sip:[email protected];user=phone;tag=619390923-1375264232667-
Call-ID: [email protected]
Max-Forwards: 27
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
Really destroying SIP dialog ‘[email protected]’ Method: ACK
Thank you for the help.