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[scott.haley at edwardj...]

I have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong?

nxdasterisk-2*CLI>
[Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted
Audio is at 18380
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.175.135:5060:
INVITE sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Contact: <sip:3145152000@192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.175.135:5060 --->
INVITE sip:913145152244@devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:192.168.122.57;lr;phase=terminating>
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: <sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68>
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: <sip:2ca13a6d@192.168.175.135;transport=udp;lr>
Record-Route: <sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr>
Record-Route: <sip:2ca13a6d@192.168.175.135;transport=udp;lr>
P-Charging-Vector: icid-value="d13ae820-caef-11e3-9b9c-6c3be5a59e68"
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones <sip:3145152000@devjones.com>
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: SM;origlocname="Asterisk-2";origsiglocname="Asterisk-2";origmedialocname="Asterisk-2";termlocname="Asterisk-2";termsiglocname="Asterisk-2";smaccounting="true"

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (27 headers 11 lines) ---
Sending to 192.168.175.135:5060 (no NAT)
Sending to 192.168.175.135:5060 (no NAT)
Using INVITE request as basis request - 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|g722), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.122.57:18380
Looking for 913145152244 in from-pstn (domain devjones.com)

<--- Reliably Transmitting (no NAT) to 192.168.175.135:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.175.135;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4;received=192.168.175.135;rport=5060
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=as119fde8b
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 23 08:20:59] NOTICE[19026][C-00000003]: chan_sip.c:25450 handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to extension '913145152244' rejected because extension not found in context 'from-pstn'.
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.175.135:5060 --->
ACK sip:913145152244@devjones.com SIP/2.0
Route: <sip:192.168.122.57;lr;phase=terminating>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=as119fde8b
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
CSeq: 102 ACK
Max-Forwards: 66
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' Method: ACK

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 403 Forbidden (Denial 1732)
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
Server: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.3.1.0.631004
Warning: 399 192.168.175.252 "Restricted Access"
To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.175.135:5060:
ACK sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
Contact: <sip:3145152000@192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0


---
[Apr 23 08:20:59] WARNING[19026][C-00000002]: chan_sip.c:22991 handle_response_invite: Received response: "Forbidden" from '"Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f'
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE)
[Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (1) Hangup
[Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts

Thanks,
Scott Haley
IS Voice Projects Team
Edward Jones Investments
Phone: 314-515-2244
Email: scott.haley@edwardjones.com (scott.haley@edwardjones.com)




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[admin at tootai.net]

Hello

Le 23/04/2014 15:36, Haley,Scott A a écrit :

[richard.seguin at mari...]

Are you using freeswitch, or just plain asterisk? I just setup a trunk between Asterisk and CM this morning, and it works great.... providing that you allow for anonymous calls.

-----Original Message-----
From: "Haley,Scott A" <scott.haley@edwardjones.com>
Sent: Wednesday, April 23, 2014 9:36am
To: "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Trunk issue

--
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http://lists.digium.com/mailman/listinfo/asterisk-usersI have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong?

nxdasterisk-2*CLI>
[Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted
Audio is at 18380
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.175.135:5060:
INVITE sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Contact: <sip:3145152000@192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.175.135:5060 --->
INVITE sip:913145152244@devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:192.168.122.57;lr;phase=terminating>
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: <sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68>
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: <sip:2ca13a6d@192.168.175.135;transport=udp;lr>
Record-Route: <sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr>
Record-Route: <sip:2ca13a6d@192.168.175.135;transport=udp;lr>
P-Charging-Vector: icid-value="d13ae820-caef-11e3-9b9c-6c3be5a59e68"
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones <sip:3145152000@devjones.com>
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: SM;origlocname="Asterisk-2";origsiglocname="Asterisk-2";origmedialocname="Asterisk-2";termlocname="Asterisk-2";termsiglocname="Asterisk-2";smaccounting="true"

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (27 headers 11 lines) ---
Sending to 192.168.175.135:5060 (no NAT)
Sending to 192.168.175.135:5060 (no NAT)
Using INVITE request as basis request - 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|g722), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.122.57:18380
Looking for 913145152244 in from-pstn (domain devjones.com)

<--- Reliably Transmitting (no NAT) to 192.168.175.135:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.175.135;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4;received=192.168.175.135;rport=5060
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=as119fde8b
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 23 08:20:59] NOTICE[19026][C-00000003]: chan_sip.c:25450 handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to extension '913145152244' rejected because extension not found in context 'from-pstn'.
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.175.135:5060 --->
ACK sip:913145152244@devjones.com SIP/2.0
Route: <sip:192.168.122.57;lr;phase=terminating>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=as119fde8b
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
CSeq: 102 ACK
Max-Forwards: 66
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' Method: ACK

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 403 Forbidden (Denial 1732)
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
Server: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.3.1.0.631004
Warning: 399 192.168.175.252 "Restricted Access"
To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.175.135:5060:
ACK sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
Contact: <sip:3145152000@192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0


---
[Apr 23 08:20:59] WARNING[19026][C-00000002]: chan_sip.c:22991 handle_response_invite: Received response: "Forbidden" from '"Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f'
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE)
[Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (1) Hangup
[Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts

Thanks,
Scott Haley
IS Voice Projects Team
Edward Jones Investments
Phone: 314-515-2244
Email: scott.haley@edwardjones.com<mailto:scott.haley@edwardjones.com>



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[scott.haley at edwardj...]

It is just plain Asterisk. I solved the original problem of it not being in the <from-pstn> context, now I am getting a rejected error I believe from the CM.

Thanks,
Scott Haley
5-2244

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of richard.seguin@marisec.ca
Sent: Wednesday, April 23, 2014 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Trunk issue

Are you using freeswitch, or just plain asterisk? I just setup a trunk between Asterisk and CM this morning, and it works great.... providing that you allow for anonymous calls.

-----Original Message-----
From: "Haley,Scott A" <scott.haley@edwardjones.com>
Sent: Wednesday, April 23, 2014 9:36am
To: "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Trunk issue

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http://lists.digium.com/mailman/listinfo/asterisk-usersI have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong?

nxdasterisk-2*CLI>
[Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.175.135:5060:
INVITE sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Contact: <sip:3145152000@192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.175.135:5060 ---> INVITE sip:913145152244@devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:192.168.122.57;lr;phase=terminating>
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: <sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68>
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: <sip:2ca13a6d@192.168.175.135;transport=udp;lr>
Record-Route: <sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr>
Record-Route: <sip:2ca13a6d@192.168.175.135;transport=udp;lr>
P-Charging-Vector: icid-value="d13ae820-caef-11e3-9b9c-6c3be5a59e68"
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones <sip:3145152000@devjones.com>
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: SM;origlocname="Asterisk-2";origsiglocname="Asterisk-2";origmedialocname="Asterisk-2";termlocname="Asterisk-2";termsiglocname="Asterisk-2";smaccounting="true"

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (27 headers 11 lines) ---
Sending to 192.168.175.135:5060 (no NAT) Sending to 192.168.175.135:5060 (no NAT) Using INVITE request as basis request - 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|g722), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.122.57:18380 Looking for 913145152244 in from-pstn (domain devjones.com)

<--- Reliably Transmitting (no NAT) to 192.168.175.135:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.175.135;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4;received=192.168.175.135;rport=5060
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=as119fde8b
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 23 08:20:59] NOTICE[19026][C-00000003]: chan_sip.c:25450 handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to extension '913145152244' rejected because extension not found in context 'from-pstn'.
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.175.135:5060 ---> ACK sip:913145152244@devjones.com SIP/2.0
Route: <sip:192.168.122.57;lr;phase=terminating>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=as119fde8b
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
CSeq: 102 ACK
Max-Forwards: 66
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' Method: ACK

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 403 Forbidden (Denial 1732)
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
Server: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.3.1.0.631004
Warning: 399 192.168.175.252 "Restricted Access"
To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.175.135:5060:
ACK sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
Contact: <sip:3145152000@192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0


---
[Apr 23 08:20:59] WARNING[19026][C-00000002]: chan_sip.c:22991 handle_response_invite: Received response: "Forbidden" from '"Edward Jones" <sip:3145152000@192.168.122.57>;tag=as4eecf94f'
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE) [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (1) Hangup [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts

Thanks,
Scott Haley
IS Voice Projects Team
Edward Jones Investments
Phone: 314-515-2244
Email: scott.haley@edwardjones.com<mailto:scott.haley@edwardjones.com>



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[scott.haley at edwardj...]

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack
[Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?

Thanks,
Scott Haley
5-2244





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[EWieling at nyigc.com]

Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack [Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?
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[scott.haley at edwardj...]

It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi->get_variable("astexten");

#$agi->answer();
#$agi->stream_file("welcome");






$agi->set_variable($dialgroup1, "$dg1value");
$agi->set_variable($dialgroup2, "$dg2value");
$agi->set_variable($vmvariable, "$vmvalue");
$agi->set_variable($timer, "$timervalue");
$agi->set_variable($branch, "$branchvalue");

Thanks,
Scott Haley
5-2244





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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack [Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?
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[asghar144 at gmail.com]

file is executable?can you show ls -l /var/lib/asterisk/agi-bin



On Mon, Apr 28, 2014 at 7:12 PM, Haley,Scott A <scott.haley@edwardjones.com (scott.haley@edwardjones.com)> wrote:

[EWieling at nyigc.com]

Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error.

If you were not running it as the same user as asterisk runs as you should still get a different error.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi->get_variable("astexten");

#$agi->answer();
#$agi->stream_file("welcome");






$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2, "$dg2value"); $agi->set_variable($vmvariable, "$vmvalue"); $agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch, "$branchvalue");

Thanks,
Scott Haley
5-2244





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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack [Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?
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[scott.haley at edwardj...]

Here is the directory listing:

[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi

Thanks,
Scott Haley
5-2244


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue


Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error.

If you were not running it as the same user as asterisk runs as you should still get a different error.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi->get_variable("astexten");

#$agi->answer();
#$agi->stream_file("welcome");






$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2, "$dg2value"); $agi->set_variable($vmvariable, "$vmvalue"); $agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch, "$branchvalue");

Thanks,
Scott Haley
5-2244





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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack [Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?
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[scott.haley at edwardj...]

One more thing. I have this exact same script working on an Asterisk 1.8 box. This is a new Asterisk 11.7 box.

Thanks,
Scott Haley
5-2244


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Here is the directory listing:

[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi

Thanks,
Scott Haley
5-2244


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue


Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error.

If you were not running it as the same user as asterisk runs as you should still get a different error.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi->get_variable("astexten");

#$agi->answer();
#$agi->stream_file("welcome");






$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2, "$dg2value"); $agi->set_variable($vmvariable, "$vmvalue"); $agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch, "$branchvalue");

Thanks,
Scott Haley
5-2244





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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack [Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?
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[asghar144 at gmail.com]

if that is the case then check again Perl Asterisk AGI.


On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A <scott.haley@edwardjones.com (scott.haley@edwardjones.com)> wrote:

[scott.haley at edwardj...]

Now I am getting Permission denied.

-- Executing [4000@phones:1] NoOp("SIP/7001-0000003a", "Starting TBS Dailer App") in new stack
-- Executing [4000@phones:2] NoOp("SIP/7001-0000003a", "4000") in new stack
-- Executing [4000@phones:3] Gosub("SIP/7001-0000003a", "tbs-utils,s,1,(4000)") in new stack
-- Executing [s@tbs-utils:1] NoOp("SIP/7001-0000003a", "Entering tbs-utils for 4000") in new stack
-- Executing [s@tbs-utils:2] Set("SIP/7001-0000003a", "DIALGROUP1=") in new stack
-- Executing [s@tbs-utils:3] Set("SIP/7001-0000003a", "DIALGROUP2=") in new stack
-- Executing [s@tbs-utils:4] Set("SIP/7001-0000003a", "VM=") in new stack
-- Executing [s@tbs-utils:5] Set("SIP/7001-0000003a", "TIMER=") in new stack
-- Executing [s@tbs-utils:6] Set("SIP/7001-0000003a", "BRANCH=") in new stack
-- Executing [s@tbs-utils:7] AGI("SIP/7001-0000003a", "tbsdial.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/tbsdial.agi
tbsdial.agi: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': Permission denied

Thanks,
Scott Haley
5-2244

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asghar Mohammad
Sent: Monday, April 28, 2014 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

if that is the case then check again Perl Asterisk AGI.


On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A <scott.haley@edwardjones.com (scott.haley@edwardjones.com)> wrote:
One more thing. I have this exact same script working on an Asterisk 1.8 box. This is a new Asterisk 11.7 box.

Thanks,
Scott Haley
5-2244


-----Original Message-----

From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Haley,Scott A

Sent: Monday, April 28, 2014 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Here is the directory listing:

[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi

Thanks,
Scott Haley
5-2244


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue


Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error.

If you were not running it as the same user as asterisk runs as you should still get a different error.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi->get_variable("astexten");

#$agi->answer();
#$agi->stream_file("welcome");






$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2, "$dg2value"); $agi->set_variable($vmvariable, "$vmvalue"); $agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch, "$branchvalue");

Thanks,
Scott Haley
5-2244





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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows:

-- Executing [s@tbs-utils:7] AGI("SIP/7002-0000001a", "tbsdial.agi") in new stack [Apr 28 12:00:05] WARNING[21812][C-0000000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user "asterisk". Why does it say the file does not exist?
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[patrick at laimbock.com]

On 28-04-14 19:49, Haley,Scott A wrote:

[scott.haley at edwardj...]

That seemed to fix it. Thanks to everyone.

Thanks,
Scott Haley
5-2244





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-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Monday, April 28, 2014 12:58 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Trunk issue

On 28-04-14 19:49, Haley,Scott A wrote: