Logging in does not initiate conference

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Logging in does not initiate conference

Postby palito » Wed Aug 01, 2007 5:35 pm

Hi. When I log into agc/vicidial.php, my extension doesn't ring. I will get the "Noone is in your session: 8600051" message. I am able, however, to dial the conference number (Session ID) from my extension, and after that I am able to connect to the MeetMe conference and work normally.

I can see messages like this in action_full.2007-xx-xx:

2007-08-01 18:24:19|0|1|
Action: Originate
Channel: SIP/6000
Context: default
Exten: 8600051
Priority: 1
Callerid: S0708011824198600051

|
2007-08-01 18:24:22|1||Response: Error
Message: Originate failed

|
2007-08-01 18:24:24|2||Response: Goodbye
Message: Thanks for all the fish.


CLI shows the following message:

NOTICE[4290]: channel.c:2511 __ast_request_and_dial: Unable to request channel SIP/6000

Asterisk version: 1.2.19
Vicidial version: 2.0.129 - Scratch installation
uname -a: Linux dialer 2.6.18-4-amd64 #1 SMP Mon Mar 26 11:36:53 CEST 2007 x86_64 GNU/Linux
Phones: SIP Softphones
Trunks: SIP

Please advise. Thank you!
palito
 
Posts: 18
Joined: Tue Jan 16, 2007 4:03 pm

Postby mflorell » Thu Aug 02, 2007 7:26 am

what is the device name of your SIP phone?
mflorell
Site Admin
 
Posts: 18387
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby palito » Thu Aug 02, 2007 11:06 am

Hi. I've defined two sip phones in sip.conf:

[6000]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=6000
secret=6000
host=dynamic
qualify=1000
context=default

[6001]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=6001
secret=6001
host=dynamic
qualify=1000
context=default
palito
 
Posts: 18
Joined: Tue Jan 16, 2007 4:03 pm

Postby vctor » Thu Aug 02, 2007 12:08 pm

I also encountered this problem when we tried a linux machine for the agent, firefox and Xlite for Linux,
vctor
 
Posts: 49
Joined: Tue Jun 13, 2006 9:32 pm

Postby palito » Thu Aug 02, 2007 6:10 pm

I think I found a hint regarding what's happening. Our * box has two NIC's. NIC 0 is hooked to our LAN, while NIC 1 is connected to our VoIP provider. Both our extensions and our provider use SIP. On further testings, I also found out that the same error shows when calling one SIP extension to another extension.

I think it must be something regarding our network configuration (maybe routing??). For the time being, we decide to go IAX for our agents, and everything works great.

I'd love to hear if anyone has further suggestions on how to tackle this... Thank you!
palito
 
Posts: 18
Joined: Tue Jan 16, 2007 4:03 pm


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