Hi, I created another callwithus.xml under outbound and it is working now.
Thanks alot. Pete On Thu, Apr 17, 2008 at 11:27 PM, UV <[EMAIL PROTECTED]> wrote: > Try: > > Sofia status gateway callwithus > > > > (maybe someone should add this to the wiki page > http://wiki.freeswitch.org/wiki/Sofia ) > ------------------------------ > > *From:* [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] *On Behalf Of *Pete Kay > *Sent:* Friday, April 18, 2008 1:08 AM > *To:* [email protected] > *Subject:* Re: [Freeswitch-users] Need help with a gateway problem > > > > Hi, > > I tried "sofia status" and it does not show my gateway, > > [EMAIL PROTECTED]> sofia status > API CALL [sofia(status)] output: > Name Type > Data State > > ================================================================================================= > 192.168.1.104 alias > default ALIASED > outbound profile sip:[EMAIL PROTECTED]:5080 > RUNNING (0) > default profile sip:[EMAIL PROTECTED]:5060 > RUNNING (0) > nat profile sip:[EMAIL PROTECTED]:5070 > RUNNING (0) > > ================================================================================================= > 3 profiles 1 alias > > I suspected maybe something is wrong with my config but I just can't > figure it out. > > Could you please help to take a look at my pastebin? > http://pastebin.freeswitch.org/4248 > > Please help me so I get get started with FS. > > Thanks, > Pete > > On Thu, Apr 17, 2008 at 10:48 PM, UV <[EMAIL PROTECTED]> wrote: > > Try this instead: > > <action application="bridge" data="sofia/gateway/callwithus/$1@ > sip.callwithus.com"/> > > > ------------------------------ > > *From:* [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] *On Behalf Of *Pete Kay > *Sent:* Thursday, April 17, 2008 11:02 PM > *To:* [email protected] > *Subject:* [Freeswitch-users] Need help with a gateway problem > > > > Hi, > > I am still not yet able to get one sip phone to call the other due to the > problem I posted. So, I used the working sip client to try to call an > outside number > by setting up a gateway. > > I tried to set up fastswitch to route call to my voip provider, I am > getting channel error but can't know why. Can anyone please give some help? > > this is in my sofia.conf.xml > > > > <gateways> > > <gateway name="callwithus"> > > <!--/// account username *required* ///--> > > <param name="username" value="226"/> > > <!--/// auth realm: *optional* same as gateway name, if blank > ///--> > > <param name="realm" value="callwithus"/> > > <!--/// account password *required* ///--> > > <param name="password" value="3336"/> > > <!--/// extension for inbound calls: *optional* same as > username, if$ <param name="extension" value="2"/> > > <!--/// proxy host: *optional* same as realm, if blank ///--> > > <param name="proxy" value="sip.callwithus.com"/> > > <!--/// expire in seconds: *optional* 3600, if blank ///--> > > <param name="expire-seconds" value="600"/> > > > > </gateway> > > </gateways> > > > > > > > > > > this is in my default dialplan > > > > <context name="home"> > > <extension name="PrintVars" continue="true"> > > <condition field="destination_number" expression="^[0-9]"> > > <action application="info"/> > > </condition> > > </extension> > > > > <extension name="PBX Extension"> > > <condition field="destination_number" expression="^(1[0-9]{2})$"> > > <!-- This will dial a registered phone at the ip or domain name you > set in$ <!-- (The % indicates it is an internal extension) > $ <action application="bridge" > data="sofia/sip/$1%${server-domain-name}"/> > > </condition> > > </extension> > > <extension name="Local Dial"> > > <condition field="destination_number" expression="^([0-9]{13})$"> > > <action application="set" data="effective_caller_id_name=John > Freeswitch"/> <action application="set" > data="effective_caller_id_number=1234567"/> > > <action application="log" data="before bridging [${1}]\n"/> > > <action application="bridge" > data="sofia/gateway/callwithus/[EMAIL PROTECTED]"$ </condition> > > </extension> > > </context> > > > > > > Get error when I dial a 13 digits number: > > > > 2008-04-18 04:26:03 [ERR] mod_sofia.c:1692 sofia_outgoing_channel() > Invalid Gateway > > 2008-04-18 04:26:03 [NOTICE] mod_sofia.c:1857 sofia_outgoing_channel() > Close Channel N/A [CS_NEW] > > 2008-04-18 04:26:03 [ERR] switch_ivr_originate.c:813 > switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: > [INVALID_NUMBER_FORMAT] > > 2008-04-18 04:26:03 [INFO] mod_dptools.c:1536 audio_bridge_function() > Originate Failed. Cause: INVALID_NUMBER_FORM AT > > > Any suggestion will be greatly appreciated. > > Thanks > Pete > > > > > > No virus found in this outgoing message. > Checked by AVG. > Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16-Apr-08 > 5:34 PM > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > No virus found in this incoming message. > Checked by AVG. > Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16-Apr-08 > 5:34 PM > > No virus found in this outgoing message. > Checked by AVG. > Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16-Apr-08 > 5:34 PM > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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