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Auto Dialing not working

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Auto Dialing not working

Postby tolen » Sun Aug 31, 2008 10:54 pm

Hi folks,

My auto dialing is not working. I have set the dialing method as ratio. The call did land on destination no but after pickingup the call it gets disconnected(after about 5/10 secs).Before the call gets disconnected I am not able to hear anything. Manual dialing is working fine. :(

Here is the asterisk console(phone number replaced by XXXXXXXXXXXX):

-- Executing [1XXXXXXXXXXXX@default:1] AGI("Local/1XXXXXXXXXXXX@default-6f60,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [1XXXXXXXXXXXX@default:2] Dial("Local/1XXXXXXXXXXXX@default-6f60,2", "SIP/MyTrunk/XXXXXXXXXXXX||tTo") in new stack
-- Called MyTrunk/XXXXXXXXXXXX
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/MyTrunk-08254bc8 is ringing
-- SIP/MyTrunk-08254bc8 is making progress passing it to Local/1XXXXXXXXXXXX@default-6f60,2
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/cc100-081f7390'
-- Executing [h@default:1] DeadAGI("SIP/cc100-081f7390", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 1XXXXXXXXXXXX, 2) exited non-zero on 'Local/1XXXXXXXXXXXX@default-6f60,2'
tolen
 
Posts: 17
Joined: Fri Aug 22, 2008 1:41 am
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Postby tolen » Mon Sep 01, 2008 12:42 am

The following is the output from sip-debug(it is some what lengthy):

-- Executing [1XXXXXXXXXXXX@default:2] Dial("Local/1XXXXXXXXXXXX@default-387a,2", "SIP/MyTrunk/XXXXXXXXXXXX||tTo") in new stack
Audio is at 192.168.5.119 port 12782
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to voip-provider:5060:
INVITE sip:XXXXXXXXXXXX@voip-provider SIP/2.0
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK267f3aa9;rport
From: "V0901104934000000076" <sip:0000000000@192.168.5.119>;tag=as035f724c
To: <sip:XXXXXXXXXXXX@voip-provider>
Contact: <sip:0000000000@192.168.5.119>
Call-ID: 5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Sep 2008 05:19:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 2667 2667 IN IP4 192.168.5.119
s=session
c=IN IP4 192.168.5.119
t=0 0
m=audio 12782 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called MyTrunk/XXXXXXXXXXXX
test*CLI>
<--- SIP read from voip-provider:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK267f3aa9;received=121.246.235.180;rport=62473
From: "V0901104934000000076" <sip:0000000000@192.168.5.119>;tag=as035f724c
To: <sip:XXXXXXXXXXXX@voip-provider>
Call-ID: 5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119
CSeq: 102 INVITE
User-Agent: sipper the rapper
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:XXXXXXXXXXXX@voip-provider:3060>
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.5.25:50166:
OPTIONS sip:cc100@192.168.5.25:50166;rinstance=995a35f3ee823f00 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK378c337f;rport
From: "asterisk" <sip:asterisk@192.168.5.119>;tag=as3440f05b
To: <sip:cc100@192.168.5.25:50166;rinstance=995a35f3ee823f00>
Contact: <sip:asterisk@192.168.5.119>
Call-ID: 21dbbeae6686e6144532479a3e30f51e@192.168.5.119
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Sep 2008 05:19:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
test*CLI>
<--- SIP read from 192.168.5.25:50166 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK378c337f;rport=5060
Contact: <sip:192.168.5.25:50166>
To: <sip:cc100@192.168.5.25:50166;rinstance=995a35f3ee823f00>;tag=555e4838
From: "asterisk"<sip:asterisk@192.168.5.119>;tag=as3440f05b
Call-ID: 21dbbeae6686e6144532479a3e30f51e@192.168.5.119
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '21dbbeae6686e6144532479a3e30f51e@192.168.5.119' Method: OPTIONS

<--- SIP read from 192.168.5.25:50166 --->



<------------->
test*CLI>
<--- SIP read from voip-provider:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK267f3aa9;received=121.246.235.180;rport=62473
From: "V0901104934000000076" <sip:0000000000@192.168.5.119>;tag=as035f724c
To: <sip:XXXXXXXXXXXX@voip-provider>;tag=as4e3dfd53
Call-ID: 5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119
CSeq: 102 INVITE
User-Agent: sipper the rapper
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:XXXXXXXXXXXX@voip-provider:3060>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 3104 3104 IN IP4 voip-provider
s=session
c=IN IP4 voip-provider
t=0 0
m=audio 13566 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port voip-provider:13566
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port voip-provider:13566
list_route: hop: <sip:XXXXXXXXXXXX@voip-provider:3060>
set_destination: Parsing <sip:XXXXXXXXXXXX@voip-provider:3060> for address/port to send to
set_destination: set destination to voip-provider, port 3060
Transmitting (no NAT) to voip-provider:3060:
ACK sip:XXXXXXXXXXXX@voip-provider:3060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.119:5060;branch=z9hG4bK552fcea9;rport
From: "V0901104934000000076" <sip:0000000000@192.168.5.119>;tag=as035f724c
To: <sip:XXXXXXXXXXXX@voip-provider>;tag=as4e3dfd53
Contact: <sip:0000000000@192.168.5.119>
Call-ID: 5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/MyTrunk-081f5958 answered Local/1XXXXXXXXXXXX@default-387a,2
> Channel Local/1XXXXXXXXXXXX@default-387a,1 was answered.
-- Executing [8365@default:1] AGI("Local/1XXXXXXXXXXXX@default-387a,1", "agi://127.0.0.1:4577/call_log") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
test*CLI>
<--- SIP read from voip-provider:3060 --->
BYE sip:0000000000@192.168.5.119 SIP/2.0
Via: SIP/2.0/UDP voip-provider:3060;branch=z9hG4bK092ad8f6;rport
From: <sip:XXXXXXXXXXXX@voip-provider>;tag=as4e3dfd53
To: "V0901104934000000076" <sip:0000000000@192.168.5.119>;tag=as035f724c
Call-ID: 5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119
CSeq: 102 BYE
User-Agent: sipper the rapper
Max-Forwards: 70
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to voip-provider : 3060 (NAT)

<--- Transmitting (NAT) to voip-provider:3060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP voip-provider:3060;branch=z9hG4bK092ad8f6;received=voip-provider;rport=3060
From: <sip:XXXXXXXXXXXX@voip-provider>;tag=as4e3dfd53
To: "V0901104934000000076" <sip:0000000000@192.168.5.119>;tag=as035f724c
Call-ID: 5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0000000000@192.168.5.119>
Content-Length: 0


<------------>
== Spawn extension (default, 1XXXXXXXXXXXX, 2) exited non-zero on 'Local/1XXXXXXXXXXXX@default-387a,2'
-- Executing [h@default:1] DeadAGI("Local/1XXXXXXXXXXXX@default-387a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----32-----16)") in new stack
Really destroying SIP dialog '5dca04815cc6f0e11e2c280a01a59c45@192.168.5.119' Method: BYE

It seems that immediately after answering the call, the voip provider sends 'BYE' . Since I am able to manuallydial the same no,I shouldnot be a codec problem. I am really confused. please help !!
tolen
 
Posts: 17
Joined: Fri Aug 22, 2008 1:41 am
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Postby mflorell » Mon Sep 01, 2008 8:17 am

Asterisk version?

vicidial version and build?

Are you using the sip-silence Playback on 8365 in your dialplan?

SIP debug doesn't do anything for me at this point.
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida
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Postby tolen » Mon Sep 01, 2008 11:47 pm

THANKS Matt for ur constant support!!!
I added sip-silence and my auto dialing is working. :P
tolen
 
Posts: 17
Joined: Fri Aug 22, 2008 1:41 am
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