Unable to request channel of asterisk

Openfire Plugins Asterisk-IM
1

I have just got the Asterisk and Wildfire working correctly, and the Asterisk-IM plugin can connect to Asterisk Manager successfully. Then i added 2 Phone Mapping entries:

Username | Device | Extension | Caller ID

user1 | SIP/user1 | 2000 | user1

user2 | SIP/user2 | 2001 | user2

When i try to place a call to user2 as user1@mycompany.com in Spark, an error message issued on Asterisk CLI console as below, then neither the caller nor the callee rung:

Destroying Call ‘‘6194936501e1023803cfcac0093f06c@127.0.0.1’’

Dec 26 22:50:24 NOTICE[30725]: channel.c:2424 __ast_request_and_dial: unable to request channel SIP/user1

In Smack debug window there are two packets:

Could anyone give me some ideas? Thanks in advance.

Regards,

feerdo

2

Do you have a sip account in asterisk sip.conf file for the channel user1?

Can you post the contents of your sip.conf? Also was your sip device successfully registered when you tried this?

–Andrew

3

Hi, Andrew, thanks for your reply.

Here is my asterisk sip.conf:

context=default

allowguest=no

bindport=5060

bindaddr=0.0.0.0

srvlookup=yes

defaultexpiry=600

videosupport=yes

relaxdtmf=yes

dtmfmode = rfc2833

subscribecontext = default

notifyringing = yes

nat=no

type=friend

context=default

host=dynamic

disallow=all

allow=gsm

allow=g723.1

allow=ulaw

allow=h263

type=friend

context=default

;canreinvite=yes

host=dynamic

disallow=all

allow=gsm

allow=g723.1

allow=ulaw

allow=h263

The account has the extension 2000, and the 2001.

Two softphones registered with Asterisk as and respectively can work no problem.

But if register as in spark, and in a softphone or in spark, neither of them works.

If register as in spark, and register as the same user in the softphone, then place a call to an arbitrary channel from spark, the softphone will be ringing.

It seems that it always try to call itself.

BTW, All clients and server are running on LAN.

Thanks for any help.

Regards,

feerdo

4

Hi

In ur sip.conf plz give the default ip to ur user1 and user2 registration

try it.

defaultip=192.168.0.82 ;Your server ip insted of this ip.

if u have any username and secret plz give that also.

secret=123

username=user1

regards bvsk

5

the same thing. It got’'s channels but no ring. Use wildfire 2.6.2 and asterisk-im 1.1. I guess I should update system. Is it safe to update from nightly builds or svn? when will be next releases with fixed bugs?

6

The svn source should be in a working state, though I have not tested it recently. The 1.1.1 release should occur in the near future but I cannot give you an exact date, it mostly depends on when I can get some bandwidth go through the issues.