Benjamin J. Bawkon
2003-Mar-26 11:35 UTC
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #198 - 7 msgs
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@mac.com>
To: asterisk-users@lists.digium.com
Reply-To: asterisk-users@lists.digium.com
Replace exten => 301,1,Dial,SIP/sip:301@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@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@192.168.0.5")
in new stack
> Called sip:301@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@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
Mike Reiling
2003-Mar-26 12:10 UTC
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #198 - 7 msgs
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@mac.com> > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > Replace exten => 301,1,Dial,SIP/sip:301@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@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@192.168.0.5") in new stack >> Called sip:301@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@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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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.