I am trying to call from one internal phone to another internal phone - but I can not get it working :(. I have an analog phone connected to a Digium TDM400P and an IP phone (identified as user:5) In directory.xml I have identified the IP phone as: <user id="5"> <params> <param name="password" value="5"/> </params> </user> In my dialplan I have: <extension name="Internal-5"> <condition field="destination_number" expression="^5$"> <action application="bridge" data="sofia/${sip_profile}/5% ${sip_profile}"/> </condition> </extension> In sophia.xml I try to make it so that sip phones have to register by having: <!--<param name="accept-blind-reg" value="true"/> --> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="true"/> When I dial '5' on the analog phone I can see that freeswitch tries to make the connection, but can not: [NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute bridge(sofia/${sip_profile}/5% ${sip_profile}) [DEBUG] switch_core_state_machine.c:158 switch_core_standard_on_execute() Expanded String bridge(sofia/192.168.0.4/5%192.168.0.4) [WARNING] mod_sofia.c:1397 sofia_outgoing_channel() Cannot locate registered user 5@192.168.0.4 Any ideas/suggestions on how to call internal sip phones ? Regards, Craig.
Is the phone actually registering to freeswitch? It appears not. Also, in the directory, is the id 5 in the domain 192.1680.0.4? You will want to look in the log before the call is made to see what happens with the register attempt (or lack of) from the phone. Mike On 8/15/07 3:40 AM, "Craig Kowald" <craig.kowald@cubemicrosystems.com> wrote:
I am trying to call from one internal phone to another internal phone - but I can not get it working :(.
I have an analog phone connected to a Digium TDM400P and an IP phone (identified as user:5)
In directory.xml I have identified the IP phone as: <user id="5"> <params> <param name="password" value="5"/> </params> </user>
In my dialplan I have: <extension name="Internal-5"> <condition field="destination_number" expression="^5$"> <action application="bridge" data="sofia/${sip_profile}/5%${sip_profile}"/> </condition> </extension>
In sophia.xml I try to make it so that sip phones have to register by having: <!--<param name="accept-blind-reg" value="true"/> --> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="true"/>
When I dial '5' on the analog phone I can see that freeswitch tries to make the connection, but can not:
[NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute bridge(sofia/${sip_profile}/5%${sip_profile}) [DEBUG] switch_core_state_machine.c:158 switch_core_standard_on_execute() Expanded String bridge(sofia/192.168.0.4/5%192.168.0.4) [WARNING] mod_sofia.c:1397 sofia_outgoing_channel() Cannot locate registered user 5@192.168.0.4
Any ideas/suggestions on how to call internal sip phones ?
Regards, Craig.
_______________________________________________ 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
The SIP profile name is an ip address? How about changing it to "foo.localdomain" On 8/15/07, Michael Jerris <mike@jerris.com> wrote:
Is the phone actually registering to freeswitch? It appears not. Also, in the directory, is the id 5 in the domain 192.1680.0.4? You will want to look in the log before the call is made to see what happens with the register attempt (or lack of) from the phone.
Mike
On 8/15/07 3:40 AM, "Craig Kowald" <craig.kowald@cubemicrosystems.com> wrote:
I am trying to call from one internal phone to another internal phone - but I can not get it working :(.
I have an analog phone connected to a Digium TDM400P and an IP phone (identified as user:5)
In directory.xml I have identified the IP phone as: <user id="5"> <params> <param name="password" value="5"/> </params> </user>
In my dialplan I have: <extension name="Internal-5"> <condition field="destination_number" expression="^5$"> <action application="bridge" data="sofia/${sip_profile}/5%${sip_profile}"/> </condition> </extension>
In sophia.xml I try to make it so that sip phones have to register by having: <!--<param name="accept-blind-reg" value="true"/> --> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="true"/>
When I dial '5' on the analog phone I can see that freeswitch tries to make the connection, but can not:
[NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute bridge(sofia/${sip_profile}/5%${sip_profile}) [DEBUG] switch_core_state_machine.c:158 switch_core_standard_on_execute() Expanded String bridge(sofia/192.168.0.4/5%192.168.0.4) [WARNING] mod_sofia.c:1397 sofia_outgoing_channel() Cannot locate registered user 5@192.168.0.4
Any ideas/suggestions on how to call internal sip phones ?
Regards, Craig. ________________________________ _______________________________________________ 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
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Well, I have no idea why, but when I booted this morning it all just worked without me changing anything :/ Sadly I did not get a log of it, but I did notice that the first time I ran freeswitch that the IP phone registered. Further, on subsequent executions of freeswitch, the IP phone does not seem to need/bother to re-register but I can make internal calls from the analog phone connected to TDM to the IP phone. Now I am off to try and sort out calling from the IP phone to the analog phone. I am getting some errors but have not looked into it yet. [ERR] mod_openzap.c:719 channel_outgoing_channel() Missing span [ERR] switch_ivr_originate.c:533 switch_ivr_originate() Cannot Create Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER To answer some questions:
Is the phone actually registering to freeswitch? It appears not.
I have noticed it register once, but not again. Is there something I should be setting/doing to force it to register every time freeswitch starts up ?
in the directory, is the id 5 in the domain 192.1680.0.4?
Yes, id 5 is in the correct domain. In directory.xml I have: <domain name="$${sip_profile}">
[change profile from ip address to domain name]
I did not do this - since it all just worked this morning. I will keep this in mind though if I have further problems. Thanks for the help. Regards, Craig. On Wed, 2007-08-15 at 11:46 -0400, Jason Garland wrote:
The SIP profile name is an ip address?
How about changing it to "foo.localdomain"
On 8/15/07, Michael Jerris <mike@jerris.com> wrote:
Is the phone actually registering to freeswitch? It appears not. Also, in the directory, is the id 5 in the domain 192.1680.0.4? You will want to look in the log before the call is made to see what happens with the register attempt (or lack of) from the phone.
Mike
On 8/15/07 3:40 AM, "Craig Kowald" <craig.kowald@cubemicrosystems.com> wrote:
I am trying to call from one internal phone to another internal phone - but I can not get it working :(.
I have an analog phone connected to a Digium TDM400P and an IP phone (identified as user:5)
In directory.xml I have identified the IP phone as: <user id="5"> <params> <param name="password" value="5"/> </params> </user>
In my dialplan I have: <extension name="Internal-5"> <condition field="destination_number" expression="^5$"> <action application="bridge" data="sofia/${sip_profile}/5%${sip_profile}"/> </condition> </extension>
In sophia.xml I try to make it so that sip phones have to register by having: <!--<param name="accept-blind-reg" value="true"/> --> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="true"/>
When I dial '5' on the analog phone I can see that freeswitch tries to make the connection, but can not:
[NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute bridge(sofia/${sip_profile}/5%${sip_profile}) [DEBUG] switch_core_state_machine.c:158 switch_core_standard_on_execute() Expanded String bridge(sofia/192.168.0.4/5%192.168.0.4) [WARNING] mod_sofia.c:1397 sofia_outgoing_channel() Cannot locate registered user 5@192.168.0.4
Any ideas/suggestions on how to call internal sip phones ?
Regards, Craig. ________________________________ _______________________________________________ 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
_______________________________________________ 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
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participants (3)
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Craig Kowald
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Jason Garland
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Michael Jerris