Bridging to external SIP gateway doesn't seems to be called...
I'm running FreeSWITCH 1.4 on Debian 7. I've hooked up my FreeSWITCH to an external SIP gateway. The following scenarios work: FS -> FS SIP GW -> FS But the following scenario does not work: FS -> SIP GW I only have one single dial plan in my setup: <include> <extension name="test-dialplan"> <condition field="destination_number" expression="^\+(\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/> </condition> </extension> </include> My users are named the same as real GSM numbers (country code and number). I would like my FS to call internally if the user exists, if not, then route it to my SIP gateway for connecting to the GSM world. As mentioned, everything works except outgoing calls to the SIP gateway. What goes wrong?
Sorry, this is my dial plan: <include> <extension name="test-dialplan"> <condition field="destination_number" expression="^\+(\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/> </condition> </extension> </include> On Sat, Sep 5, 2015 at 11:03 AM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I'm running FreeSWITCH 1.4 on Debian 7.
I've hooked up my FreeSWITCH to an external SIP gateway.
The following scenarios work:
FS -> FS SIP GW -> FS
But the following scenario does not work:
FS -> SIP GW
I only have one single dial plan in my setup:
<include>
<extension name="test-dialplan">
<condition field="destination_number" expression="^\+(\d{3,20})$">
<action application="export" data="dialed_extension=$1"/>
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
<action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/>
<action application="bridge" data="user/${dialed_extension}@ ${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/>
</condition>
</extension>
</include>
My users are named the same as real GSM numbers (country code and number).
I would like my FS to call internally if the user exists, if not, then route it to my SIP gateway for connecting to the GSM world.
As mentioned, everything works except outgoing calls to the SIP gateway.
What goes wrong?
Hello everyone... Apparently it was an issue with my username from my SIP provider. They've given my an e-mail address as username, but authentication from FreeSWITCH failed with this. When they changed my username to a regular all a-z characters only, everything worked. Apoligies for this, it wasn't a FS issue. On Sat, Sep 5, 2015 at 11:11 AM, Michael Nielsen <mic.niel84@gmail.com> wrote:
Sorry, this is my dial plan:
<include>
<extension name="test-dialplan">
<condition field="destination_number" expression="^\+(\d{3,20})$">
<action application="export" data="dialed_extension=$1"/>
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
<action application="bridge" data="user/${dialed_extension}@ ${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/>
</condition>
</extension>
</include>
On Sat, Sep 5, 2015 at 11:03 AM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I'm running FreeSWITCH 1.4 on Debian 7.
I've hooked up my FreeSWITCH to an external SIP gateway.
The following scenarios work:
FS -> FS SIP GW -> FS
But the following scenario does not work:
FS -> SIP GW
I only have one single dial plan in my setup:
<include>
<extension name="test-dialplan">
<condition field="destination_number" expression="^\+(\d{3,20})$">
<action application="export" data="dialed_extension=$1"/>
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
<action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/>
<action application="bridge" data="user/${dialed_extension}@ ${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/>
</condition>
</extension>
</include>
My users are named the same as real GSM numbers (country code and number).
I would like my FS to call internally if the user exists, if not, then route it to my SIP gateway for connecting to the GSM world.
As mentioned, everything works except outgoing calls to the SIP gateway.
What goes wrong?
participants (1)
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Michael Nielsen