IP Phone is not registering


#1

Hi Friends,

     I am configuring the Cisco IP Phone 7600 with asterisk server, as extension 121. But that IP phone not able to register.

IP phone is sending REQUEST packet, but Asterisk server is not able to send 100 TRYING packet and then, i am getting 401 Unauthenticated packet from Asterisk server.

When i check the log messages from Asterisk server, i got the following messages.

[Sep 27 13:31:39] WARNING[5103] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
170 [Sep 27 13:31:48] WARNING[5104] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
171 [Sep 27 13:32:16] WARNING[4813] chan_sip.c: Maximum retries exceeded on transmission 001a2f53-cae10004-23cb5ee4-72160 3b3@192.168.1.12 for seqno 101 (Non-critical Response) – See wiki.asterisk.org/wiki/display/AST/SIP+Retransm issions

The following is the sip.conf and extension.conf files.
[121]
context=sipphone
type=friend
host=dynamic
username=121
secret=121

[122]
context=sipphone
type=friend
host=dynamic
username=122
secret=122

[277]
context=sipphone
type=friend
host=dynamic
username=277
secret=277

extensions.conf
[sipphone]
exten => 101,1,Dial(SIP/jis)
exten => 100,1,Dial(SIP/prem)
exten => 102,1,Dial(SIP/sonu)
exten => 103,1,Dial(SIP/anush)
exten => 121,1,Dial(SIP/121)
exten => 122,1,Dial(SIP/122)
exten => 111,1,Dial(SIP/111)
exten => 222,1,Dial(SIP/222)
exten => 333,1,Dial(SIP/333)
exten => 444,1,Dial(SIP/444)
exten => 277,1,Dial(SIP/277)
exten => 278,1,Dial(SIP/278)
exten => 279,1,Dial(SIP/279)
exten => 280,1,Dial(SIP/280)
exten => 104,1,Dial(SIP/mathew)
exten => 365,1,Dial(SIP/365)

Please any help me how to do this.

Regards
Nagavardhan.


#2

REQUEST is not a standard SIP method. You really need to provide the actual SIP trace.

Unable to create channel just means the phone isn’t registered, so it doesn’t no where to send the outgoing call.

The critical packet message implies that Asterisk is sending somewhere but not getting any responses. This can be a NAT issue, but, for example, I know one soft phone that fails to respond to a re-invite, when it should actively reject it.


#3

Now i enabled the NAT in my SIP.cnf file. Now i can call to SJPhone. But i can’t able to call to Cisco IP phone.

What might be the wrong that i have done. Can you please explain me.

Regards,
Nagavardhan.