Hi, Here's a problem I posted about on the IRC some time ago - I said I'd do some traces and put them in pastebin, which I've done here: http://pastebin.freeswitch.org/13970 A brief description of the problem and the calls involved in the pastebin. Incoming calls are delivered to the Freeswitch from a SIP trunking provider who are also used to route outgoing calls. This is not via a gateway. The SIP trunking provider insists that for outgoing calls, the data in the "From" field in the INVITE message contains a PSTN number that they have 'associated' with the SIP trunking service, e.g. From: "Zoltan" <sip:02031375892@79.125.110.116> is OK. From: "Zoltan" <sip:anonymous@79.125.110.116> would be rejected by them, as would From: "Zoltan" <sip:01344299800@79.125.110.116> as 01344299800 is 'someone elses' number, not one of theirs that they have allocated to us. The first call in the pastebin shows a Polycom phone making an outgoing call that works / is allowed as the effective_caller_id_number is set to an "allowed" number 02031375892. The second call, starting at line 871 shows a call originating in the PSTN from number 01344299800 to the Polycom. The SIP provider delivers the call to the FreeSwitch which in turn sends the call to the Polycom. The Polycom has been set to divert all incoming calls to number 07770380800 and replies with a "SIP/2.0 302 Moved Temporarily" with the forwarded to number as the contact. This in turn causes the FreeSwitch to generate an INVITE message which is sent to the trunk provider with: From: "01344299800" <sip:01344299800@79.125.110.116> and To: <sip:07770380800@79.125.4.174> Since 01344299800 is not an "allowed" number according to the provider, the call is rejected. Putting aside the argument for now of whether the SIP provider should be carrying out this rather brutal form of "CLI screening" (the data in the 'From' field appears as the A Party Number / CLI once the call hits the PSTN / mobile networks), is there any way to modify the behaviour of this call so that in response to the "SIP/2.0 302 Moved Temporarily" message, the INVITE which is sent out has the number from the effective_caller_id_number variable for the forwarding extension, rather than the number from the "From" field of the incoming invite? Hope this makes sense, Mike
Mike, All of our redirect handling information is covered here: http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect If I were you I would play around with the sip-related variables like sip_to_uri and sip_to_user when handling the 302 redirect from the Polycom. Let us know what you find out. -MC On Wed, Sep 22, 2010 at 5:59 AM, Mike <motosota@gmail.com> wrote:
Hi,
Here's a problem I posted about on the IRC some time ago - I said I'd do some traces and put them in pastebin, which I've done here:
http://pastebin.freeswitch.org/13970
A brief description of the problem and the calls involved in the pastebin.
Incoming calls are delivered to the Freeswitch from a SIP trunking provider who are also used to route outgoing calls. This is not via a gateway.
The SIP trunking provider insists that for outgoing calls, the data in the "From" field in the INVITE message contains a PSTN number that they have 'associated' with the SIP trunking service, e.g. From: "Zoltan" <sip:02031375892@79.125.110.116<sip%3A02031375892@79.125.110.116>> is OK. From: "Zoltan" <sip:anonymous@79.125.110.116<sip%3Aanonymous@79.125.110.116>> would be rejected by them, as would From: "Zoltan" <sip:01344299800@79.125.110.116<sip%3A01344299800@79.125.110.116>> as 01344299800 is 'someone elses' number, not one of theirs that they have allocated to us.
The first call in the pastebin shows a Polycom phone making an outgoing call that works / is allowed as the effective_caller_id_number is set to an "allowed" number 02031375892.
The second call, starting at line 871 shows a call originating in the PSTN from number 01344299800 to the Polycom. The SIP provider delivers the call to the FreeSwitch which in turn sends the call to the Polycom. The Polycom has been set to divert all incoming calls to number 07770380800 and replies with a "SIP/2.0 302 Moved Temporarily" with the forwarded to number as the contact. This in turn causes the FreeSwitch to generate an INVITE message which is sent to the trunk provider with: From: "01344299800" <sip:01344299800@79.125.110.116<sip%3A01344299800@79.125.110.116>> and To: <sip:07770380800@79.125.4.174 <sip%3A07770380800@79.125.4.174>>
Since 01344299800 is not an "allowed" number according to the provider, the call is rejected.
Putting aside the argument for now of whether the SIP provider should be carrying out this rather brutal form of "CLI screening" (the data in the 'From' field appears as the A Party Number / CLI once the call hits the PSTN / mobile networks), is there any way to modify the behaviour of this call so that in response to the "SIP/2.0 302 Moved Temporarily" message, the INVITE which is sent out has the number from the effective_caller_id_number variable for the forwarding extension, rather than the number from the "From" field of the incoming invite?
Hope this makes sense,
Mike
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participants (2)
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Michael Collins
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Mike