[Spce-user] Sip INVITE Callee ID Number

[Deon Vermeulen]

Hi

Anyone able to assist me with this?


Thanks

Kind Regards
Deon Vermeulen

Sent from my iPhone

On May 1, 2013, at 8:43, Deon Vermeulen <vermeulen.deon at gmail.com> wrote:

> 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
> 
> 
> 
>> <postbox-contact.jpg>	Jon Bonilla (Manwe)	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? 
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>> <postbox-contact.jpg>	Deon Vermeulen	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 172.16.0.250:5070;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 172.16.0.250;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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