vicidial.org

by **jojo1031** » Mon Oct 12, 2009 2:18 pm

I tried to add a DID using the web-based DID configuration on the admin.php but when I did a test call to newly added DID it gave a prompt extension not found. I double checked the extension-vicidial.conf and I can't find the DID that I added.

Any advice?

Admin.php version and build:
VERSION: 2.0.5-174
BUILD: 90522-0506

Regards,

Jojo

by **mflorell** » Mon Oct 12, 2009 5:44 pm

How did you install ViciDial?

Was this an upgrade from an earlier version?

do you have the trunkinbound context?

what kind of trunks are you using?

by **jojo1031** » Mon Oct 12, 2009 7:02 pm

I installed vicidial from scratch.

Its a new install not an upgrade.

I have a trunkinbound context on my extensions.conf

Im using SIP trunking.

Thanks Matt.

by **williamconley** » Mon Oct 12, 2009 11:24 pm

vicidial doesn't place the extensions inthe conf file, everything is handled via databasing in the agi script instead.

what does your cli say when the call comes in? does it go to the agi?

by **jojo1031** » Tue Oct 13, 2009 12:13 am

It shows nothing on the cli and a promt "the person you are calling is unavailble" will be heard over the phone. When I tried to manually add the DID in the extensions.conf, everything is Ok.

Below is the cli output with sip debug enabled:

<-- SIP read from 119.111.121.7:33638:
INVITE sip:8005555555@202.57.38.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.88:33638;branch=z9hG4bK-d87543-393aeb3354213d7d-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:802@119.111.121.7:33638>
To: "8005555555"<sip:8005555555@202.57.38.98>
From: "802"<sip:802@202.57.38.98>;tag=695f5761
Call-ID: MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 449

v=0
o=- 7 2 IN IP4 192.168.100.88
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.100.88
t=0 0
m=audio 23568 RTP/AVP 107 100 106 6 0 105 18 3 5 101
a=alt:1 1 : u3ayO06n fmtMDMcb 192.168.100.88 23568
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:E880E7873AB049FA8D0D15A2A1903917

--- (12 headers 16 lines) ---
Using INVITE request as basis request - MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
Sending to 192.168.100.88 : 33638 (NAT)
Reliably Transmitting (NAT) to 119.111.121.7:33638:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.88:33638;branch=z9hG4bK-d87543-393aeb3354213d7d-1--d87543-;received=119.111.121.7;rport=33638
From: "802"<sip:802@202.57.38.98>;tag=695f5761
To: "8005555555"<sip:8005555555@202.57.38.98>;tag=as287f72c2
Call-ID: MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e4e7219"
Content-Length: 0

---
Scheduling destruction of call 'MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.' in 15000 ms
Found user '802'
vicidial*CLI>
<-- SIP read from 119.111.121.7:33638:
ACK sip:8005555555@202.57.38.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.88:33638;branch=z9hG4bK-d87543-393aeb3354213d7d-1--d87543-;rport
To: "8005555555"<sip:8005555555@202.57.38.98>;tag=as287f72c2
From: "802"<sip:802@202.57.38.98>;tag=695f5761
Call-ID: MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
CSeq: 1 ACK
Content-Length: 0

--- (7 headers 0 lines) ---
vicidial*CLI>
<-- SIP read from 119.111.121.7:33638:
INVITE sip:8005555555@202.57.38.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.88:33638;branch=z9hG4bK-d87543-a0401c2727440869-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:802@119.111.121.7:33638>
To: "8005555555"<sip:8005555555@202.57.38.98>
From: "802"<sip:802@202.57.38.98>;tag=695f5761
Call-ID: MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="802",realm="asterisk",nonce="0e4e7219",uri="sip:8005555555@202.57.38.98",response="cf2cc136c2ffa0be3a0148661953e968",algorithm=MD5
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 449

v=0
o=- 7 2 IN IP4 192.168.100.88
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.100.88
t=0 0
m=audio 23568 RTP/AVP 107 100 106 6 0 105 18 3 5 101
a=alt:1 1 : u3ayO06n fmtMDMcb 192.168.100.88 23568
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:E880E7873AB049FA8D0D15A2A1903917

--- (13 headers 16 lines) ---
Using INVITE request as basis request - MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
Sending to 192.168.100.88 : 33638 (NAT)
Found user '802'
Found RTP audio format 107
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 192.168.100.88:23568
Found description format BV32
Found description format SPEEX
Found description format SPEEX-FEC
Found description format SPEEX-FEC
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x326 (gsm|ulaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8005555555 in default (domain 202.57.38.98)
Reliably Transmitting (NAT) to 119.111.121.7:33638:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.100.88:33638;branch=z9hG4bK-d87543-a0401c2727440869-1--d87543-;received=119.111.121.7;rport=33638
From: "802"<sip:802@202.57.38.98>;tag=695f5761
To: "8005555555"<sip:8005555555@202.57.38.98>;tag=as287f72c2
Call-ID: MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
vicidial*CLI>
<-- SIP read from 119.111.121.7:33638:
ACK sip:8005555555@202.57.38.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.88:33638;branch=z9hG4bK-d87543-a0401c2727440869-1--d87543-;rport
To: "8005555555"<sip:8005555555@202.57.38.98>;tag=as287f72c2
From: "802"<sip:802@202.57.38.98>;tag=695f5761
Call-ID: MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.
CSeq: 2 ACK
Content-Length: 0

--- (7 headers 0 lines) ---
Destroying call 'MWU0ZTYyMDZlMjRkNWJjMDIzYTY4ZTIwMjFjMDI4NzU.'
vicidial*CLI>
<-- SIP read from 119.111.121.7:33638:

--- (0 headers 1 lines) ---
vicidial*CLI>
<-- SIP read from 119.111.121.7:33638:

by **ronsrussell** » Tue Oct 13, 2009 3:17 am

I had the same problem with an old system that got upgraded to the latest release. My solution was to hard code the DID's in extensions.conf. On this system the dynamic DID's worked for a couple of weeks and then failed. This system gets rebooted every night.

Operating system SuSE Linux 10.3
CPU load averages 0.15 (1 min) 0.13 (5 mins) 0.10 (15 mins)
Real memory 7.67 GB total, 586.04 MB used
Virtual memory 2.01 GB total, 88 kB used
Local disk space 132.49 GB total, 82.03 GB used

VICIdial
VERSION: 2.2.0-215
BUILD: 90908-1207

by **williamconley** » Tue Oct 13, 2009 1:48 pm

hm.

IF you add to the extensions.conf, and the CLI does show the inbound call (and the call works), but when you add via the GUI the call does not appear in the CLI at all, there may be an issue with your install.

the default DID inbound route should take the call and pass it to "no service", which is visible in the CLI, even if the DID does not exist in your setup.

try comparing it to a DEMO cd (Vicibox DEMO). and see what the difference is between your setup and their setup.

by **jojo1031** » Tue Oct 13, 2009 5:07 pm

I actually installed vicibox to compare with my server side by side and I didn't find anything wrong. The system I installed was working perfectly (outbound and inbound calling with 100% recording) though the web gui for DID is the only thing that is not working. What i did for now is hard code the DID in to extensions.conf.

Any advice?

Regards,

Jojo

by **williamconley** » Tue Oct 13, 2009 5:22 pm

when you did your comparison, did you bring a did call in through the demo box?

I realize that his represents a certain amount of (seemingly wasted) work, considering that you'll have to undo it almost immediately. but finding out if it works inbound DID to the demo would give you a serious troubleshooting opportunity. two boxes with the same software sitting next to each other ...

by **oshonubi** » Thu Oct 15, 2009 1:24 pm

Hi All,

I am not too sure that the DID at the web interface works at least I have tried it without success with the Zap interface. I would like to confirm from anyone who had had a success with it to help.

Is there any one with successful DID configuration at the front end?

by **williamconley** » Thu Oct 15, 2009 5:51 pm

used it many times. on several boxes. be sure you have your zap set to context 'trunkinbound' or it won't FIND the built-in did router and land in the right agi script.

by **oshonubi** » Fri Oct 16, 2009 4:45 am

Hi William,

Thanks for the response. However from all indication, the instruction on the manual, precisely tutorial c focuses on using ingroup as the DID route. In this case, I don't think extension is a reference rather it is the group_id. Upon using the in-group as the DID route, all the configurations are not saved at the front end, rather we had to go to the backend, that is extensions.conf.

Is there extension configuration in the in-group route rather than the in_group ID

vicidial.org

Web-based DID configuration not working?

Web-based DID configuration not working?

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