how to get rid of second P-Asserted-Identity?
When receiving a call from PSTN and forwarding it to another PSTN number and setting P-Asserted-Identity header, I found that 2 P-Asserted-Identity headers are present in the INVITE message. As the target provider only checks the first one, the call is denied. The first P-Asserted-Identity header is from the the incoming call. The second is set in our dialplan. Is there achance to drop the first header part? Best regards Peter Here's the dialplan: <extension name="Forward to Net"> <condition field="destination_number" expression="^\S*$"> <action application="set" data="effective_caller_id_number=02x1204xxxxx"/> <action application="set" data="effective_caller_id_name=02x1204xxxxx"/> <action application="export"><![CDATA[sip_h_P-Asserted-Identity=<sip:02x1204xxxxx@my.domain.de>]]></action> <action application="export"><![CDATA[sip_h_P-Preferred-Identity=<sip:${caller_id_number}@my.domain.de>]]></action> <action application="bridge" data="sofia/external/0162xxxxxxxxxxxxxx@provider.domain"/> </condition> </extension> Here is the corresponding INVITE. INVITE sip:0162xxxxxxxxxxxxxx@sip1.my.domain.de SIP/2.0. Via: SIP/2.0/UDP 82.xxx.xx.1x3:5080;rport;branch=z9hG4bKjD7UvXB8XF5jD. Max-Forwards: 29. From: "02x139xxxxx" <sip:02x139xxxxx@82.xxx.xx.1x3>;tag=U4v0ypBB1X78F. To: <sip:0162xxxxxxxxxxxxxx@sip1.my.domain.de>. Call-ID: 1cdc72b9-f7ea-122d-e685-001ec9b9dd2d. CSeq: 132448209 INVITE. Contact: <sip:mod_sofia@82.xxx.xx.1x3:5080>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15434M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. X-FS-Support: update_display. P-Asserted-Identity: "02x139xxxxx" <sip:02x139xxxxx@82.xxx.xx.1x3>. (from incoing call) P-Asserted-Identity: <sip:02x1204xxxxx@my.domain.de>. (set in dialplan) P-Preferred-Identity: <sip:02x139xxxxx@my.domain.de>. . v=0. o=FreeSWITCH 1277106526 1277106527 IN IP4 82.xxx.xx.1x3. s=FreeSWITCH. c=IN IP4 82.xxx.xx.1x3. t=0 0. m=audio 26564 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20.
Hello, I finally found the "sip_cid_type=none" setting. Best regards Peter Peter P GMX schrieb:
When receiving a call from PSTN and forwarding it to another PSTN number and setting P-Asserted-Identity header, I found that 2 P-Asserted-Identity headers are present in the INVITE message. As the target provider only checks the first one, the call is denied.
The first P-Asserted-Identity header is from the the incoming call. The second is set in our dialplan. Is there achance to drop the first header part?
Best regards Peter
Here's the dialplan: <extension name="Forward to Net"> <condition field="destination_number" expression="^\S*$"> <action application="set" data="effective_caller_id_number=02x1204xxxxx"/> <action application="set" data="effective_caller_id_name=02x1204xxxxx"/> <action application="export"><![CDATA[sip_h_P-Asserted-Identity=<sip:02x1204xxxxx@my.domain.de>]]></action> <action application="export"><![CDATA[sip_h_P-Preferred-Identity=<sip:${caller_id_number}@my.domain.de>]]></action> <action application="bridge" data="sofia/external/0162xxxxxxxxxxxxxx@provider.domain"/> </condition> </extension>
Here is the corresponding INVITE. INVITE sip:0162xxxxxxxxxxxxxx@sip1.my.domain.de SIP/2.0. Via: SIP/2.0/UDP 82.xxx.xx.1x3:5080;rport;branch=z9hG4bKjD7UvXB8XF5jD. Max-Forwards: 29. From: "02x139xxxxx" <sip:02x139xxxxx@82.xxx.xx.1x3>;tag=U4v0ypBB1X78F. To: <sip:0162xxxxxxxxxxxxxx@sip1.my.domain.de>. Call-ID: 1cdc72b9-f7ea-122d-e685-001ec9b9dd2d. CSeq: 132448209 INVITE. Contact: <sip:mod_sofia@82.xxx.xx.1x3:5080>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15434M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. X-FS-Support: update_display. P-Asserted-Identity: "02x139xxxxx" <sip:02x139xxxxx@82.xxx.xx.1x3>. (from incoing call) P-Asserted-Identity: <sip:02x1204xxxxx@my.domain.de>. (set in dialplan) P-Preferred-Identity: <sip:02x139xxxxx@my.domain.de>. . v=0. o=FreeSWITCH 1277106526 1277106527 IN IP4 82.xxx.xx.1x3. s=FreeSWITCH. c=IN IP4 82.xxx.xx.1x3. t=0 0. m=audio 26564 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20.
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Peter P GMX