[Spce-user] Sip INVITE Callee ID Number
Deon Vermeulen
vermeulen.deon at gmail.com
Wed May 1 03:43:32 EDT 2013
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?
> _______________________________________________
> Spce-user mailing list
> 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
> 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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