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/$1@callwithus"$ </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
Try this instead: <action application="bridge" data="sofia/gateway/callwithus/$1@HYPERLINK "http://sip.callwithus.com"sip.callwithus.com"/> _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pete Kay Sent: Thursday, April 17, 2008 11:02 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Need help with a gateway problem Hi, I am still not yet able to get one sip phone to call the other due to the problem I posted. So, I used the working sip client to try to call an outside number by setting up a gateway. I tried to set up fastswitch to route call to my voip provider, I am getting channel error but can't know why. Can anyone please give some help? this is in my sofia.conf.xml <gateways> <gateway name="callwithus"> <!--/// account username *required* ///--> <param name="username" value="226"/> <!--/// auth realm: *optional* same as gateway name, if blank ///--> <param name="realm" value="callwithus"/> <!--/// account password *required* ///--> <param name="password" value="3336"/> <!--/// extension for inbound calls: *optional* same as username, if$ <param name="extension" value="2"/> <!--/// proxy host: *optional* same as realm, if blank ///--> <param name="proxy" value="HYPERLINK "http://sip.callwithus.com"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/$1@callwithus"$ </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
Hi, I tried "sofia status" and it does not show my gateway, freeswitch@ser> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= 192.168.1.104 alias default ALIASED outbound profile sip:mod_sofia@58.251.75.228:5080 RUNNING (0) default profile sip:mod_sofia@192.168.1.104:5060 RUNNING (0) nat profile sip:mod_sofia@58.251.75.228:5070 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 <uv@talknet.com.au> wrote:
Try this instead:
<action application="bridge" data="sofia/gateway/callwithus/$1@ sip.callwithus.com"/>
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Pete Kay *Sent:* Thursday, April 17, 2008 11:02 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Need help with a gateway problem
Hi,
I am still not yet able to get one sip phone to call the other due to the problem I posted. So, I used the working sip client to try to call an outside number by setting up a gateway.
I tried to set up fastswitch to route call to my voip provider, I am getting channel error but can't know why. Can anyone please give some help?
this is in my sofia.conf.xml
<gateways>
<gateway name="callwithus">
<!--/// account username *required* ///-->
<param name="username" value="226"/>
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<param name="realm" value="callwithus"/>
<!--/// account password *required* ///-->
<param name="password" value="3336"/>
<!--/// extension for inbound calls: *optional* same as username, if$ <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/$1@callwithus"$ </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 Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Try: Sofia status gateway callwithus (maybe someone should add this to the wiki page HYPERLINK "http://wiki.freeswitch.org/wiki/Sofia"http://wiki.freeswitch.org/wiki/Sofia ) _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pete Kay Sent: Friday, April 18, 2008 1:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Need help with a gateway problem Hi, I tried "sofia status" and it does not show my gateway, freeswitch@ser> sofia status API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== HYPERLINK "http://192.168.1.104"192.168.1.104 alias default ALIASED outbound profile HYPERLINK "http://sip:mod_sofia@58.251.75.228:5080"sip:mod_sofia@58.251.75.228:5080 RUNNING (0) default profile HYPERLINK "http://sip:mod_sofia@192.168.1.104:5060"sip:mod_sofia@192.168.1.104:5060 RUNNING (0) nat profile HYPERLINK "http://sip:mod_sofia@58.251.75.228:5070"sip:mod_sofia@58.251.75.228:5070 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? HYPERLINK "http://pastebin.freeswitch.org/4248"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 <HYPERLINK "mailto:uv@talknet.com.au"uv@talknet.com.au> wrote: Try this instead: <action application="bridge" data="sofia/gateway/callwithus/$1@HYPERLINK "http://sip.callwithus.com" \nsip.callwithus.com"/> _____ From: HYPERLINK "mailto:freeswitch-users-bounces@lists.freeswitch.org" \nfreeswitch-users-bounces@lists.freeswitch.org [mailto:HYPERLINK "mailto:freeswitch-users-bounces@lists.freeswitch.org" \nfreeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pete Kay Sent: Thursday, April 17, 2008 11:02 PM To: HYPERLINK "mailto:freeswitch-users@lists.freeswitch.org" \nfreeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Need help with a gateway problem Hi, I am still not yet able to get one sip phone to call the other due to the problem I posted. So, I used the working sip client to try to call an outside number by setting up a gateway. I tried to set up fastswitch to route call to my voip provider, I am getting channel error but can't know why. Can anyone please give some help? this is in my sofia.conf.xml <gateways> <gateway name="callwithus"> <!--/// account username *required* ///--> <param name="username" value="226"/> <!--/// auth realm: *optional* same as gateway name, if blank ///--> <param name="realm" value="callwithus"/> <!--/// account password *required* ///--> <param name="password" value="3336"/> <!--/// extension for inbound calls: *optional* same as username, if$ <param name="extension" value="2"/> <!--/// proxy host: *optional* same as realm, if blank ///--> <param name="proxy" value="HYPERLINK "http://sip.callwithus.com" \nsip.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/$1@callwithus"$ </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 HYPERLINK "mailto:Freeswitch-users@lists.freeswitch.org"Freeswitch-users@lists.freeswi tch.org HYPERLINK "http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" \nhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:HYPERLINK "http://lists.freeswitch.org/mailman/options/freeswitch-users" \nhttp://lists.freeswitch.org/mailman/options/freeswitch-users HYPERLINK "http://www.freeswitch.org" \nhttp://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
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 <uv@talknet.com.au> wrote:
Try:
Sofia status gateway callwithus
(maybe someone should add this to the wiki page http://wiki.freeswitch.org/wiki/Sofia ) ------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Pete Kay *Sent:* Friday, April 18, 2008 1:08 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Need help with a gateway problem
Hi,
I tried "sofia status" and it does not show my gateway,
freeswitch@ser> sofia status API CALL [sofia(status)] output: Name Type Data State
================================================================================================= 192.168.1.104 alias default ALIASED outbound profile sip:mod_sofia@58.251.75.228:5080 RUNNING (0) default profile sip:mod_sofia@192.168.1.104:5060 RUNNING (0) nat profile sip:mod_sofia@58.251.75.228:5070 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 <uv@talknet.com.au> wrote:
Try this instead:
<action application="bridge" data="sofia/gateway/callwithus/$1@ sip.callwithus.com"/>
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Pete Kay *Sent:* Thursday, April 17, 2008 11:02 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Need help with a gateway problem
Hi,
I am still not yet able to get one sip phone to call the other due to the problem I posted. So, I used the working sip client to try to call an outside number by setting up a gateway.
I tried to set up fastswitch to route call to my voip provider, I am getting channel error but can't know why. Can anyone please give some help?
this is in my sofia.conf.xml
<gateways>
<gateway name="callwithus">
<!--/// account username *required* ///-->
<param name="username" value="226"/>
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<param name="realm" value="callwithus"/>
<!--/// account password *required* ///-->
<param name="password" value="3336"/>
<!--/// extension for inbound calls: *optional* same as username, if$ <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/$1@callwithus"$ </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 Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Well I don't personally put them in the config like that. I put then in the directory. You can check out the .noload files I put in SVN to demonstrate that. In addition you should put this on the profile called outbound you do not want a provider on a profile you auth-calls on usually. /b On Apr 17, 2008, at 8:01 AM, Pete Kay wrote:
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/ $1@callwithus"$ </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
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participants (3)
-
Brian West
-
Pete Kay
-
UV