Trunk name is displayed instead of SIP header

[Kasper-1973]

Hi experts,

I have two OmniPCX Enterprice systems (Rev.11.1) up and running. They both have SIP clients and the interconnection is by the use of SIP ext. Gateways.

I have configured a trunk to be used with the gateway and it all works out fine.

The problem is when a call is made between the systems, then the called phone displays the trunk name instead of the information found in the sip header.

I know it's a simple issue, I just don't know where to do the "adjustment".

Kind regards
Kasper

[tot3nkopf]

Calling number is in From or P-Asserted field?
Play with P-Asserted-ID in calling number yes/no

Is the trunk group declared as private or public? Is it declared as ABCF or ISDN all countries? Are you routing using routing number or ARS between the two systems?

[vad]

Configured something in Translator/ ext. dialing plan/ ext. calback translation?

[hojzle]

Hi to all,
we also have this kind of situation and I would kindly ask you for your help.

Let me first explain the situation and configuration.

The customer use OXE with SWR R11.1 that was recently upgraded from SWR7.1. Also OT was implemented.

So, they have about ten remote locations and there are installed telef. systems such as OXO, OXE, 3CX, Asterisk, CallManager, Ascom... and all of them are connected to OXE with SIP trunk. After the upgrade on OXE site the customer started too report about issues with calling number or name presentation for inbound calls. The called party on OXE see only SIP trunk group name. We are not sure if this problem existed before the upgrade was made and at this point it doesn't matter(maybe the customer simply become more interested on call information since they know the there was an upgrade, it happens lot of times ).

We done the traces and found out that:
- if there is INVITE with name/number info in FROM header - the call is presented ok(trace 1)
- if there is INVITE with name/number info in FROM header and also P-Asserted-ID field, OXE show only SIP trunk group name!!!
- if there is INVITE only with P-Asserted-ID field or P-Prefered-ID OXE show only SIP trunk group name!!!(trace 2)

On OXE site configuration is done with:
- SIP ABC-F trunk group
- ARS is used for call routing

x Trunk Group ID : 60 x
x Trunk Group Type + T2 x
x Trunk Group Name : SIP x
x Q931 signal variant + ABC-F x
x T2 Specificity + SIP x
x Homogenous network for direct RTP + NO x
x Public Network Category : 31 x
x DDI transcoding + False x
x Can support UUS in SETUP + True x
x Associated Ext SIP gateway : -1 x

- SIP EXTGW Nb:5

lqConsult/Modify: External Gatewaysqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqk
x x
x Node Number (reserved) : 1 x
x Instance (reserved) : 1 x
x Instance : 5 x
x x
x Gateway Name : SIP 3CX x
x Remote domain : 192.168.0.4 x
x PCS IP address : ----------------------------------------------- x
x Port number : 5060 x
x Transport type + UDP x
x Belonging domain : -------------------------------------------------- x
x Registration Id : -------------------------------------------------- x
x Registration ID in P_Asserted + False x
x Registration timer : 0 x
x Outbound Proxy : -------------------------------------------------- x
x Supervision timer : 0 x
x Trunk group number : 60 x
x Pool Number : -1 x
x Outgoing realm : -------------------------------------------------- x
x Outgoing username : -------------------------------------------------- x
x x
x Outgoing Password : -------------------- x
x Confirm : -------------------- x
x x
x Incoming username : -------------------------------------------------- x
x x
x Incoming Password : -------------------- x
x Confirm : -------------------- x
x x
x RFC 3325 supported by the distant + True x
x DNS type + DNS A x
x First DNS IP Address : ----------------------------------------------- x
x Second DNS IP Address : ----------------------------------------------- x
x SDP in 18x + True x
x Minimal authentication method + None x
x INFO method for remote extension + False x
x To EMS + False x
x SRTP + RTP only x
x Ignore inactive/black hole + False x
x Contact with IP address + False x
x Dynamic Payload type for dtmf : 97 x
x Outbound Calls 100 REL + Supported x
x Incoming Calls 100 REL + Not Requested x
x Gateway type + Standard type x
x Re-Trans No. for REGISTER/OPTIONS : 2 x
x P-Asserted-ID in Calling Number + False x
x Trusted P-Asserted-ID header + False x
x Diversion Info to provide via + History Info x
x Proxy identification on IP address + False x
x Outbound calls only + False x
x SDP relay on Ext. Call Fwd + Default x
x SDP Transparency Override + False x
x RFC 5009 supported / Outbound call + Not Supported x
x Nonce caching activation + NO x
x FAX Procedure Type + T38 only x
x DNS SRV/Call retry on busy server : 0 x
x Unattended Transfer for RSI + NO x
x Redirection functionality + NO x
x Attended Transfer + NO x
x Send BYE on REFER + YES x
x Support UTF8 characters set + NO x
x CSTA User-to-User supported + NO x
x Trusted From header + True x
x Support Re-invite without SDP + True x
x Type of codec negotiation + Default x
x x

I have also set parameter Via Header_ Inbound Calls Routing + True (System/OSP/SIP). I've tryed to change parameter P-Asserted-ID in Calling Number + False to True(restart od sip and also reboot) but it didn't change the behavior. I've done it also with SIP ISDN configuration and the behavior is/was the same.



Trace 1 - OK

1450691763 -> Mon Dec 21 10:56:03 2015 RECEIVE MESSAGE FROM NETWORK (172.19.12.50:5060 [UDP])
----------------------utf8-----------------------
INVITE sip:2010@10.64.105.5 SIP/2.0

Via: SIP/2.0/UDP 172.19.12.50:5060;branch=z9hG4bK3cb8ab0f

Max-Forwards: 70

From: "Amira Soro" <sip:5656@172.19.12.50>;tag=as0109c664

To: <sip:2010@10.64.105.5>

Contact: <sip:5656@172.19.12.50:5060>

Call-ID: 143924de6416354c2e1c936441eace9d@172.19.12.50:5060

CSeq: 102 INVITE

User-Agent: FPBX-2.11.0(11.13.0)

Date: Mon, 21 Dec 2015 10:00:29 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 257



v=0

o=root 947241910 947241910 IN IP4 172.19.12.50

s=Asterisk PBX 11.13.0

c=IN IP4 172.19.12.50

t=0 0

m=audio 11726 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

-------------------------------------------------



Trace 2 - NOT OK
-------------------------------------------------
Mon Dec 21 10:58:21 2015 RECEIVE MESSAGE FROM NETWORK (172.17.11.253:5060 [UDP])
----------------------utf8-----------------------
INVITE sip:2010@10.64.105.5;user=phone SIP/2.0

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE

Supported: 100rel,from-change,timer

User-Agent: OxO_GW_710/143.001

Session-Expires: 43200

P-Asserted-Identity: <sip:4155@172.17.11.253;user=phone>

To: <sip:2010@10.64.105.5;user=phone>

From: <sip:4155@172.17.11.253;user=phone>;tag=193638a89968b46dcb80241880db2c32

Contact: <sip:4155@172.17.11.253;transport=UDP;user=phone>

Content-Type: application/sdp

Call-ID: e94f180e62b7809e0f6b5357f921507e@172.17.11.253

CSeq: 331961806 INVITE

Via: SIP/2.0/UDP 172.17.11.253;rport;branch=z9hG4bKb2216e9300a6f77faf06b5a1c93dd4cb

Max-Forwards: 70

Content-Length: 217



v=0

o=default 1450694277 1450694277 IN IP4 172.17.11.253

s=-

c=IN IP4 172.17.11.253

t=0 0

m=audio 32000 RTP/AVP 18 106

a=sendrecv

a=rtpmap:106 telephone-event/8000

a=fmtp:106 0-15

a=ptime:20

a=maxptime:120

-------------------------------------------------



In privious posts also Ext.Callback was mentioned and here is the configuration of the table:

lq[ 6 ] Instances: Ext.Callback Translation Rulesqk
x x
x -> 61 x
x 70 x
x 74 x
x 75 x
x B x
x DEF x
x x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqj

lqConsult/Modify: Ext.Callback Translation Rulesqqqqqk
x x
x Node Number (reserved) : 1 x
x Instance (reserved) : 1 x
x Instance (reserved) : 1 x
x External Callback Table : 0 x
x Basic Number : DEF x
x x
x Nb.Digits To Be Removed : 0 x
x Digits To Add : 0 x
x x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqj


lqConsult/Modify: Ext.Callback Translation Rulesqqqqqk
x x
x Node Number (reserved) : 1 x
x Instance (reserved) : 1 x
x Instance (reserved) : 1 x
x External Callback Table : 0 x
x Basic Number : B x
x x
x Nb.Digits To Be Removed : 1 x
x Digits To Add : 0 x
x x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqj


If there are some informations you need, please tell me. Thank you.

Best regards

Marko

[tot3nkopf]

You have one external gateway per link. Play with the external gateway parameters stated above.
The SIP implementation is different for 3CX, Cisco, etc. therefore you need to configure where the name/number will be extracted for for each link.

[sadim]

Hi,

Set "Trusted P-Asserted-ID header" to True.
Do not Forget to restart the sipmotor

Regards

[hojzle]

Hi tot3nkopf and sadim,

thank you for your responce.

I know about the differences in sipextgw settings for different vendors. I've mainly played with configuration of sipextgw for oxo and 3cx. I've tryed various configurations but I didn't succedd with configuration for number or name to be shown on phoneset display. I work with customer database in our lab and I have all posibilities to play around(restart of sipmotor, reboot of cpu...). Next step I'll take is to setup an empty database and do the tests again. Maybe it vent something wrong with database translation in the upgrade procedure....

I'll let you know the results I'll get. If you have any other idea, please don't hesitate to tell me... I'm desperate

Have a nice Christmas and New Year holidays

Best regards

Marko

[samet]

hi,
You can do as follows.
-> Shelf-> Descend hierarchy -> Descend hierarchy ->Spec. Customer Features Parameters-> Review/Modify ->Display trunk group name+false

[morio]

could not find this parameter. by the way isnt this in the system parameter - attendant features?

[samet]

soryy under the system i wrote wrong.
system->Descend hierarchy -> Descend hierarchy ->Spec. Customer Features Parameters-> Review/Modify ->Display trunk group name+false