[Spce-user] Sip INVITE Callee ID Number

Thu May 2 04:58:40 EDT 2013

Hi Andreas

Thank you very much for your email.

We also had a public holiday yesterday, but I was unaware it was in 
Europe as well.

I'll look into that.

Kind Regards
Deon

> Andreas Granig <mailto:agranig at sipwise.com>
> May 2, 2013 8:58 AM
> Hi,
>
> On 05/02/2013 09:27 AM, Deon Vermeulen wrote:
>> I'm not sure if Jon is not in the office or too occupied at the moment.
>
> Well, it's a community mailing-list with best effort support, and 
> yesterday was public holiday in pretty much all of Europe...
>
>> Any possibility you could assist me in this matter?
>
> If you want to change the To-header, there is no way to do it out of 
> the box. And I'm always quite puzzled why on earth in 2013 SIP 
> termination carriers still don't understand why they should leave the 
> To header alone and just not check it. Anyways...
>
> Copy your proxy.cfg.tt2 to proxy.cfg.customtt.tt2 and search for 
> "P-Called-Party-ID". This is the header we pass on to Sems, which in 
> turn uses it to replace the To header with.
>
> So if you need the display part as well, you need to manually adapt 
> the assignment of this header in your proxy configuration.
>
> Andreas
> Deon Vermeulen <mailto:vermeulen.deon at gmail.com>
> May 2, 2013 8:27 AM
> Hi Andreas
>
> I'm not sure if Jon is not in the office or too occupied at the moment.
>
> Any possibility you could assist me in this matter?
>
>
>
> Thanks
>
>
> Kind Regards
> Deon
>
> Deon Vermeulen <mailto:vermeulen.deon at gmail.com>
> May 1, 2013 8:43 AM
> Hi Jon
>
> The current INIVTE format is as follows:
>
> Max-Forwards: 70
> Record-Route: 
> <sip:10.222.0.250:6090;r2=on;lr=on;ftag=75669547-518052F2000CD143-E2039700;ngcplb=yes>
> Record-Route: 
> <sip:127.0.0.1:6090;r2=on;lr=on;ftag=75669547-518052F2000CD143-E2039700;ngcplb=yes>
> Via: SIP/2.0/UDP 
> 10.222.0.250:6090;branch=z9hG4bK12a4.4fd66e7061dd8a4bdfb0ee967839ab64.0
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK83MlJabZ;rport=5080
> From: "+44837000000" 
> <sip:+44837000000 at 192.168.0.250>;tag=75669547-518052F2000CD143-E2039700
> To: <sip:+44113001417 at 172.16.0.250>
> CSeq: 10 INVITE
> Call-ID: 00f190d0-2c90-1231-d390-68b5996c937e_b2b-1
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
> REGISTER, NOTIFY
> Supported: precondition, path, replaces
> P-Asserted-Identity: <sip:+44837000000 at 192.168.0.250>
> Content-Type: application/sdp
> Content-Length: 246
> Contact: <sip:ngcp-lb at 10.222.0.250:6090;ngcpct='sip:127.0.0.1:5080'>
>
>
>
> The required INVITE format should be:
>
> INVITE sip:+44113001417 at 172.16.0.250:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.250:6090;branch=z9hG4bKe2gvrn1008t0jnkjl2n0.1
> Call-ID: 00f190d0-2c90-1231-d390-68b5996c937e_b2b-1
> From: "+44837000000" 
> <sip:+44837000000 at 172.16.0.250;user=phone>;tag=75669547-518052F2000CD143-E2039700
> To: "+44113001417" <sip:+44113001417 at 172.16.0.250;user=phone>
> CSeq: 1 INVITE
> Contact: <sip:+44837000000 at 192.168.0.250:6090;transport=udp;user=phone>
> Allow: 
> INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
> Supported: 100rel
> P-Asserted-Identity: <tel:+44837000000 at 192.168.0.250:6090>
> Max-Forwards: 70
> Content-Length: 377
> Content-Type: application/sdp
>
>
> The current setup is as follows:
>
> - Features/Services disabled
>         * Asterisk Uninstalled, voice mail disabled.
>         * rateomat
>         * conference
>         * vsc
>         * Under ww_admin
>             - billing_features: 0
>             - conference_features: 0
>             - vsc_features: 0
>             - voicemail_features: 0
>         * Under ww_csc
>             - display_account_info: 0
>             - main_menu - All are set to 0
>             - payment_features: 0
>             - Removed all language options and set default_language: en
>             - default_uri: ''
>
> - SEMS
>         * sbc:
>             - calltimer_enable: 'yes'
>             - session_timer:
>                 - enable: 'yes'
>
> -    PROXY
>              - allow_peer_relay: 'yes'
>
>
> - PEERINGS:
>           * From Wholesale:
>               - force_outbound_calls_to_peer: 'yes'
>           * To Carrier:
>               - force_outbound_calls_to_peer: 'yes'
>               - sst_enable: 'yes'
>               - sst_expires: 300
>               - sst_min_timer: 90
>               - sst_max_timer: 7200
>               - outbound_from_user: User-Provided-Number
>               - outbound_from_display: Network-provided-Number
>               - outbound_pai_user: Network-provided-Number
>
>
> I've looked all over kamailio/sems configurations but can't find what 
> I'm looking for to "fix" the INVITE format especially the Session 
> timer type.
>
> I keep getting a 500 from the upstream carrier.
>
>
> Any assistance with this will be appreciated.
>
>
> Thank you very much
>
> Kind Regards
> Deon
>
>
>
> Jon Bonilla (Manwe) <mailto:jbonilla at sipwise.com>
> April 26, 2013 3:04 PM
> El Fri, 26 Apr 2013 14:54:56 +0100
>
>
>
>
> What's the To header you get and what's the one you want to achieve? 
> And what's
> exactly your carrier's requirement?
> _______________________________________________
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> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> Deon Vermeulen <mailto:vermeulen.deon at gmail.com>
> April 26, 2013 2:54 PM
> Hi How can I go about to format the SIP TO: Header in the INVITE as 
> follows.
>
>
> INVITE sip:1234567890 at 192.168.200.153;user=phone SIP/2.0
>
> Via: SIP/2.0/UDP 
> 192.168.200.130:5070;branch=z9hG4bK75b28f842f76d9c9e299a80c0;rport;X-DispCookie=1013;X-HwTRC=259446;X-DptMsg=1405
>
> Route: <sip:192.168.200.153:5060;transport=udp;lr>
>
> Call-ID: bb4bcb34ec7041237fbfc1e63f687744 at 10.222.0.250
>
> From: 
> "0987654321"<sip:0987654321 at 192.168.200.130;transport=udp;user=phone>;tag=fd64e901-CC-1013-TRC-259446
>
> To:"1234567890"<sip:1234567890 at 192.168.200.153;transport=udp;user=phone>
>
>
> This is a requirement by upstream carrier.
>
>
> Thanks again
>
>
> Kind Regards
> Deon
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