vicidial.org

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!

what is the device name of your SIP phone?

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

I also encountered this problem when we tried a linux machine for the agent, firefox and Xlite for Linux,

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!