[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #198 - 7 msgs

Mike Reiling miker at mac.com
Wed Mar 26 12:10:13 MST 2003


SIP/301 not SIP.301.


--Mike

On Wednesday, March 26, 2003, at 10:35  AM, Benjamin J. Bawkon wrote:

> Replacing that line as directed gives this as console debug:
>
> 	Executing DIAL("OSS/dsp", "SIP.301") in new stack
> 	NOTICE[155663]: File app_dial.c, Line 449 (dial_exec): Unable to
> create channel of type 'SIP'
> 	Everyone is busy at this time
> 	WARNING[155663]: File pbx.c, Line 1268 (ast_pbx_run): Timeout,
> but no rule 't' in context 'local'
>
> Could it be my sip client?  I'm using SJphone currently.
>
>
>
> Subject: Re: [Asterisk-Users] Dialing SIP
> From: Mike Reiling <miker at mac.com>
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
> Replace exten => 301,1,Dial,SIP/sip:301 at 192.168.0.5 with:
> 	exten => 301,1,Dial,SIP/301
>
> --Mike
>
> On Wednesday, March 26, 2003, at 08:40  AM, Benjamin J. Bawkon wrote:
>
>> Im really starting to get the hang of Asterisk, however, I still have
>> one issue...
>>
>> My SIP Client can dial other extensions just fine, but no extension
> can
>> ring the Sip client...
>>
>> Here is the pertinent info:
>> SIP.CONF,
>> [general]
>> port = 5060
>> bindaddr = 192.168.0.5 	;ip of asterisk server
>> context = default
>>
>> [301]
>> username=301
>> context=local
>> type=friend
>> secret=test
>> insecure=yes
>> host=dynamic
>>
>> ----------------------------------------
>> EXTENSIONS.CONF
>>
>> [local]
>> exten => _1XX,1,Dial,ZAP/1/BYEXTENSION
>> exten => 301,1,Dial,SIP/sip:301 at 192.168.0.5	; again, ip of * server
>>
>> blah blah blah below this..
>>
>> ----------------------------------------
>> Console Debug:
>>
>> When 301 is Dialed:
>>
>> --Executing Dial("OSS/dsp", "SIP/sip:301 at 192.168.0.5") in new stack
>> Called sip:301 at 192.168.0.5
>> Got SIP response 482 "Loop Detected" back from 192.168.0.5
>> No one is available to answer at this time
>> WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no
>> rule 't' in context 'local'
>>
>> ----------------------------------------
>> Problem is, the SIP Client never rang.....
>>
>> Now...If I change the extensions.conf to read:
>> Exten => 301,1,Dial,SIP/sip:301 at 192.168.0.109
>>
>> Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address
>> to
>> the sip client machine...It will change occasionally...
>>
>> Any Ideas?  Thanks!
>> Ben Bawkon
>
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Mike Reiling
Systems & Network Administrator
SoftCoin, Inc.
2000 Sierra Point Parkway
Brisbane, CA  94005
650-624-3869 - P
650-624-3899 - F

It might look like I'm doing nothing, but at the cellular level I'm 
really quite busy.




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