Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name="caller-id-in-from" value="true" in gw definition, and even <action application="set" data="effective_caller_id_number=07539600000"/> in the dial plan to no avail Can anyone shed some light on this? Regards, This is a sip debug sent 1382 bytes to udp/[XXX.XXX.XXX.XXX]:5060 at 21:45:17.404609: INVITE sip:07539000000@mygw.net SIP/2.0 Via: SIP/2.0/UDP 87.238.75.206:5080;rport;branch=z9hG4bKm2UveeUvtNc5D Max-Forwards: 69 From: "Extension 1000" <sip:me@mygw.net;transport=udp>;tag=j6m5U7g4F3XNa To: <sip:07539600000@gk.magrathea.net> Call-ID: 1e8c5d1d-6e71-122c-fdba-001a4b0a67ca CSeq: 110816823 INVITE Contact: <sip:me@myextIP:5080;transport=udp> Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Proxy-Authorization: Digest username="me", realm="mygw.net", nonce="498b5f19b089db1bf9d13b2c83f45407048ede9b", algorithm=MD5, uri="sip:07539600000@mygw.net", response="03dbbeef1387a11c7100965dcfd01052" Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 311 P-Key-Flags: keys="3"
Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote:
Try as I might, I cannot seem to get caller ID passed to the external sip gateway
This GW happily processes caller id from Asterisk
If tried adding param name="caller-id-in-from" value="true" in gw definition, and even
<action application="set" data="effective_caller_id_number=07539600000"/> in the dial plan to no avail
Can anyone shed some light on this?
Regards,
No good, I tried <action application="set" data="effective_caller_id_number=07539600000"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/> But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not? Regards ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 22:29 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote: Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name="caller-id-in-from" value="true" in gw definition, and even <action application="set" data="effective_caller_id_number=07539600000"/> in the dial plan to no avail Can anyone shed some light on this? Regards,
Nik, Ignore me... set should have worked... You're using caller-id-in-from let me look closer at this. /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote:
No good, I tried
<action application="set" data="effective_caller_id_number=07539600000"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/>
But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not?
Regards
Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote:
No good, I tried
<action application="set" data="effective_caller_id_number=07539600000"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/>
But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not?
Regards
Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers <extension name="dial_Nik_mobile"> <condition field="destination_number" expression="^(0\d+)$"> <action application="set" data="effective_caller_id_number=0------00000000006"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/> <action application="bridge" data="sofia/gateway/Mag/$1"/> </condition> </extension> Mag.xml <include> <gateway name="Mag"> <param name="username" value="me"/> <param name="realm" value="mygw"/> <param name="password" value="pass"/> <param name="proxy" value="GWaddress"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> </gateway> </include> ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 22:54 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: No good, I tried <action application="set" data="effective_caller_id_number=07539600000"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/> But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not? Regards
I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote:
Dial plan is as per default setup with the addition of the following. To be honest, and I’m no SIP guru, I can’t see the caller- id being set in the sip headers
<extension name="dial_Nik_mobile"> <condition field="destination_number" expression="^(0\d+)$"> <action application="set" data="effective_caller_id_number=0------00000000006"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/> <action application="bridge" data="sofia/gateway/Mag/$1"/> </condition> </extension>
Mag.xml
<include> <gateway name="Mag"> <param name="username" value="me"/> <param name="realm" value="mygw"/> <param name="password" value="pass"/> <param name="proxy" value="GWaddress"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> </gateway> </include>
Yes, I'll report back tomorrow, Regards ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers <extension name="dial_Nik_mobile"> <condition field="destination_number" expression="^(0\d+)$"> <action application="set" data="effective_caller_id_number=0------00000000006"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/> <action application="bridge" data="sofia/gateway/Mag/$1"/> </condition> </extension> Mag.xml <include> <gateway name="Mag"> <param name="username" value="me"/> <param name="realm" value="mygw"/> <param name="password" value="pass"/> <param name="proxy" value="GWaddress"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> </gateway> </include>
Ok, It's now working as expected, looks like I had something odd set in the phone's peer definition. I'll try and back track to see what I was doing wrong, but what ever it was, was preventing the caller id from being sent in the INVITE. Regards ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 05 February 2009 23:26 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Yes, I'll report back tomorrow, Regards ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers <extension name="dial_Nik_mobile"> <condition field="destination_number" expression="^(0\d+)$"> <action application="set" data="effective_caller_id_number=0------00000000006"/> <action application="export" data="effective_caller_id_number"/> <action application="privacy" data="no"/> <action application="bridge" data="sofia/gateway/Mag/$1"/> </condition> </extension> Mag.xml <include> <gateway name="Mag"> <param name="username" value="me"/> <param name="realm" value="mygw"/> <param name="password" value="pass"/> <param name="proxy" value="GWaddress"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> </gateway> </include>
participants (2)
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Brian West
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Nik Middleton