[Asterisk-Users] Dialing SIP

[Mike Reiling]

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.