Sip CID Type NOT work

[Baji Zsolt]

I need to send P-Preferred-Identity for SIP provider. But what evher i set up for gateway in Sip CID Type, freeswitch always send Remote-Party-ID. My provider block calls if I send Remote-Party-ID, now the Caller ID In From for this gateway is disabled.

What is the problem? Tried to turn on Caller ID In From =true and Sip CID Type = pid, result always same, i see in debug Remote-Party-ID and not the P-Preferred-Identity or P-Asserted-Identity.

The system is up to date, updated one week ago.

 

[TheOperator]

Are you on stable or master ?
From what I remember the Sip CID Type setting was buggy in some versions of FusionPBX/Freeswitch.
I got it working by setting the following in Advanced -> Switch Variables -> Defaults:
Name: sip_cid_type
Value: pid
Enabled: True

Then flush memcache and restart your profiles.

 

[Baji Zsolt]

Don't work.....

same result:

INVITE sip:***@sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 10.251.35.2:5080;rport;branch=z9hG4bKg00QaQgj606re
Max-Forwards: 69
From: "Outbound Call" <sip:***@10.251.35.2>;tag=D5m4c2m73ZS9g
To: <sip:00498992289355@sipconnect.sipgate.de>
Call-ID: 0cbf1b69-5abd-1236-7e98-00133b0feb8d
CSeq: 116253377 INVITE
Contact: <sip:gw+571e057e-e772-4b05-843b-941bc15fea1e@10.251.35.2:5080;transport=udp;gw=571e057e-e772-4b05-843b-941bc15fea1e>
Call-Info: <sip:10.251.35.2>;answer-after=0
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 289
X-accountcode: 10.251.35.2
X-FS-Support: update_display,send_info
P-Asserted-Identity: "Outbound Call" <sip:***@10.251.35.2>

 

[TheOperator]

According to the data you posted above you are sending the "P-Asserted-Identity" header. What might be causing a headache is the "Outbound Call" text. At least my providers here in Germany (Telekom, Easybell & Dusnet) want to see my number there (formatted as +49xxxxxxxx). Check your "Outbound Caller ID Name" & "Outbound Caller ID Number" settings on the extension used.

 

[Baji Zsolt]

The problem is not the "Outbound Call", I have tried +4XXXX and other combinations wich works with other providerr, and this is onlu the name, number is after in <sip:****>. Sipgate says that he need P-Preferred-Identity... I never get P-Preferred-Identity in headers, whatever of combinations tried....

 

[TheOperator]

In your original post you stated that either P-Preferred-Identity or P-Asserted-Identity is acceptable by your provider.
As far as I know, Freeswitch only supports Remote-Party-ID (RPID) and P-Asserted-Identity (PID) nativly. Usually providers support one of them.

It is possible to strip headers and ad custom headers to outbound calls - take a look at the freeswitch documentation: https://wiki.freeswitch.org/wiki/Sofia-SIP
Thus it should be possible to add P-Preferred-Identity via entries in the outbound route XML.

 

[DigitalDaz]

In outbound route edit it and add an action before bridge:

Action: set
Data: sip_h_P-Preferred-Identity: <sip:XXXXXXXXXXXXXXXX@sipconnect.sipgate.de>

In gateway set sip cid type to none, all the usual flush memcache restart freeswitch etc.

 

[raitokun]

In outbound route edit it and add an action before bridge:

Action: set
Data: sip_h_P-Preferred-Identity: <sip:XXXXXXXXXXXXXXXX@sipconnect.sipgate.de>

In gateway set sip cid type to none, all the usual flush memcache restart freeswitch etc.

Sorry, If i want to disable RPID before bridge call , how can do it bro ?