[OpenSIPS-Users] Caller id

Miha miha at softnet.si
Fri Jan 10 10:10:53 CET 2014


@Razvan tnx for you answer.

I tried with remove_hf() but I this did not work. I guess due to failure 
route as request that enter failure route does not have RPID or p-asserted.

SIP/2.0 302 Moved Temporarily.
To: <sip:38618108753 at opensips>;tag=77cc99cb150aeeefi0.
From: "38618108758" <sip:38618108758 at RTP_IP>;tag=tarm9Ucep73Um.
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
Contact: <sip:018108756 at opensips>.
Diversion: "38618108753" <sip:38618108753 at opensips>;reason=unconditional.
Server: Linksys/SPA922-6.1.5(a).
Content-Length: 0.

br
miha


Dne 1/9/2014 8:37 AM, piše Răzvan Crainea:
> Hi, Miha!
>
> remove_hf() is the function you should use. According to your 
> scenario, you should remove the RPID and PAI in the failure route. Are 
> you sure that code is reached for the second INVITE?
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 01/07/2014 03:52 PM, Miha wrote:
>> Hi,
>>
>> how can I remove/not send RPID and P-asserted identity?
>>
>> Opensips sends to UAC (with RPID and p-asserted), uac sends back 302
>> request and that due to 302 I am doing new invite with opensips but in
>> this invite I can see RPID and p-asserted.
>>
>> I am trying to remove it with remove_hf() but this does not works.
>>
>> How can I deal with this issue.
>>
>> here is a sip trace:
>>
>> U opensips:5060 -> UAC_PUBLIC_IP:13647
>> INVITE sip:38618108753 at UAC_PUBLIC_IP:13647 SIP/2.0.
>> Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
>> Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
>> Via: SIP/2.0/UDP
>> RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
>> Max-Forwards: 67.
>> From: "38618108758" <sip:38618108758 at RTP_IP>;tag=tarm9Ucep73Um.
>> To: <sip:38618108753 at opensips>.
>> Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
>> CSeq: 54214780 INVITE.
>> Contact: <sip:mod_sofia at RTP_IP:5080>.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.2.17+git~20131230T193020Z~52377f0f65~64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 207.
>> X-Call_id: 56c99a91-d0ef0551 at 172.31.1.103.
>> X-FS-Support: update_display,send_info.
>> P-Asserted-Identity: <sip:0038618108758 at RTP_IP;user=phone>.
>> Remote-Party-ID: 0038618108758
>> <sip:0038618108758 at RTP_IP>;party=calling;id-type=subscriber;privacy=off;screen=yes. 
>>
>> .
>> v=0.
>> o=FreeSWITCH 1389081737 1389081738 IN IP4 RTP_IP.
>> s=FreeSWITCH.
>> c=IN IP4 RTP_IP.
>> t=0 0.
>> m=audio 19952 RTP/AVP 0 8 9 101 13.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:30.
>>
>>
>> U UAC_PUBLIC_IP:13647 -> opensips:5060
>> SIP/2.0 302 Moved Temporarily.
>> To: <sip:38618108753 at opensips>;tag=77cc99cb150aeeefi0.
>> From: "38618108758" <sip:38618108758 at RTP_IP>;tag=tarm9Ucep73Um.
>> Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
>> CSeq: 54214780 INVITE.
>> Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
>> Via: SIP/2.0/UDP
>> RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
>> Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
>> Contact: <sip:018108756 at opensips>.
>> Diversion: "38618108753" 
>> <sip:38618108753 at opensips>;reason=unconditional.
>> Server: Linksys/SPA922-6.1.5(a).
>> Content-Length: 0.
>>
>>
>> U opensips:5060 -> RTP_IP:5060
>> INVITE sip:30238618108756 at opensips SIP/2.0.
>> Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
>> Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.1.
>> Via: SIP/2.0/UDP
>> RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
>> Max-Forwards: 67.
>> From: "38618108758" <sip:38618108758 at RTP_IP>;tag=tarm9Ucep73Um.
>> To: <sip:38618108753 at opensips>.
>> Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
>> CSeq: 54214780 INVITE.
>> Contact: <sip:mod_sofia at RTP_IP:5080>.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.2.17+git~20131230T193020Z~52377f0f65~64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 207.
>> X-Call_id: 56c99a91-d0ef0551 at 172.31.1.103.
>> X-FS-Support: update_display,send_info.
>> P-Asserted-Identity: <sip:0038618108758 at RTP_IP;user=phone>.
>> Remote-Party-ID: 0038618108758
>> <sip:0038618108758 at RTP_IP>;party=calling;id-type=subscriber;privacy=off;screen=yes. 
>>
>> Moved: 38618108753.
>> .
>> v=0.
>> o=FreeSWITCH 1389081737 1389081738 IN IP4 RTP_IP.
>> s=FreeSWITCH.
>> c=IN IP4 RTP_IP.
>> t=0 0.
>> m=audio 19952 RTP/AVP 0 8 9 101 13.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:30.
>>
>> tnx!
>>
>> miha
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
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>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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