CallerID to outbound trunks on Sipnet.ru

[dsen]

Hello,
Originally the issue was reported to International Russian forum but 3CX person (archie) recommended to leave bug report there.

The issue:
There is incoming trunk from Sipnet.ru.
This trunk works perfect. CallerID is working from PSTN.
There is second trunk to external IVR. For connection to external IVR we need to send INVITE without authorization to dedicated IP.
It doesn't ask authorization and doesn't request authorization. Need to send Invite with CallerID and CalledID to dedicated IP.
I created outgoing trunk to this IP. I had to put AuthID and External Number because it is mandatory. It was random value.
Next I configured Provider Outbound Parameters:
Contact: User Part - CallerNum
From user: -CallerNum

Next I routed incoming trunk from PSTN to 'Forward to Outside Number.
And routed this Outside Number to outgoing trunk to IVR.

The calls is going. Voice is going. But IVR gets CallerID as AuthID on outgoing trunk.
We discussed this question with Archie and it looks as bug.
I checked this via Wireshark. 3CX sends ALWAYS CallerID to outgoing trunk - AuthID which I configured for outgoing trunk.
Advanced Settings don't work.
Could you look?
Thanks so much.

 

[dsen]

Sorry, addition. I have 3CX 7.1 beta.
The same. For version 6, 7.0 is the same.

 

[dsen]

It looks as moderator changed Subject. It is wrong.
Right subject is "3CX doesn't send right CallerID to outbound trunk."

Please change.

 

[[email protected]]

We have noticed a similar problem.

We (and our customers) want to use the Outbound Caller ID from Ext. to show when we call via a trunk. The only way to achieve this is (I think) to set the "From: Display Name" and "From: User Part" SIP fields to "CallerName". This works but is strange.

 

[N_G]

We will recheck this setting.

The subject was changed because it did not correctly reflect the issue. 3CX does send caller ID on outbound trunks. It has to do with configuration for a specific VoIP provider.

 

[dsen]

I cheched some version. Always the same.
For outbound calls 3CX sends "AuthID" instead "CallerID".
It looks as it is for SIP trunks.

 

[dsen]

I configured for "From:" Caller Name instead CallerID. As somebody recommended me above.
I got empty field. "From: <sip:192.168.2.108". Without anything at all.
When I configured "From: CallerNum'. I got AuthID instead CallerID. Like From: <sip:[email protected]> where 123 is info which I put to Auth field for this trunk.

 

[archie]

Ok, I had a chance to look deeper in this issue.

The fact is that on outbound calls through trunk $CallerNum contains the external number of the trunk as specified in configuration. It has a reason: consider analogue PBX - when you call mobile from your office extension 100 you will see the land-line number of your office, not "100", right? So we're doing during routing to external line. It was not intended to pass internal extension's number outside.

As an alternative you can really use $CallerName. By default it will be populated with Display-name of call's originator, i.e. you can set it on your phone. Also it could be overriden at PBX using 'Outbound caller ID' option of extension's settings. If it is not set neither on the phone nor at OutCID - you will get it blank, exactly as you did.

For outbound calls 3CX sends "AuthID" instead "CallerID".

To be precisely correct $CallerNum in that case equals to External Number ($LineNumber), not AuthID. If you make them different - you will notice it.

 

[dsen]

Hi,
I wrote before:
"I configured for "From:" Caller Name instead CallerID. I got empty field. "From: <sip:192.168.2.108". Without anything at all."
Caller Name didn't help.
It doesn't work.
And I can't put fixed CallerID as well. I need dinamic CallerID from incoming trunk.
I have tandem call. Need to transfer incoming CallerID to Outgoing SIP trunk.
I understand it is not easy question.
Discussion:
1. I have Sip incoming call over incoming SIP trunk which is rerouted to SIP adjunct over outgoing SIP trunk. It is not trunk outside switch. It is outgoing trunk to SIP Endpoint.
2. A lot of PBX can do this. Cheap PBX can do as well. Why 3CX can't do this?
3. All normal PBX can modify CallerID and re-send inside private network for example.

I understand it looks as not bug. It looks as new feature. But I believe improvement of this feature will improve 3CX.
I have experience in telecommunication more that 12 years. I know a lot of switches. And only some worst switches can't do this.
I don't believe that it is hard to do so.
Archie, We can have a call. If you would like to discuss this of course. Send me your number by e-mail or you can call me. Doesn't matter.

 

[SY]

Could you please specify your requirements?
it is simple:
1. "My PBX receives following SIP INVITE request from my VoIP provider/PSTN gateway". (sample)
2. "I need to generate following INVITE to my VoIP/PSTN trunk" (sample which is based on 1.)

Thanks

 

[[email protected]]

Thanks Archie for your reply.

But I am still not able to do what I want with our trunks.

Example:

My name configured in phone is "Jim Oskarsson", my Outbound Caller ID set on 3CX extension is "46501395302".
In the outgoing Invite I want this to look like: From: "Jim Oskarsson" <sip:[email protected]>.

This is important for two reasons:
- It´s nice to be able to show the callers real name when making direct SIP calls and the callers phone number when making calls to PSTN.
- We love the feature "Append Queue Name to Caller ID" but can't use it with our current settings. When somebode makes a call that go through a queue or ring group and the call is routed out through the trunk, the queue name gets appended to the From: User part of Invite which makes the provider refuse the call due to "Illegal A-number" (we need to place a valid phone number without anything appended in the user part).

How can we achieve this?

 

[dsen]

Hi All,
I see I touched very important issue. :lol:

My original requerements are the following:
1. Incoming trunk 1 from PSTN over SIP has CallerID.
2. I reroute this call to outging SIP trunk 2 to my IVR.
3. I want to get incoming CallerID (trunk1) to outgoing CallerID to trunk 2.
Very easy. Now it doesn't work.

 

[dsen]

Stepan,
Example:
I have incoming INVITE.

Request-Line: INVITE sip:[email protected]:18973;rinstance=c4786e7da226d379 SIP/2.0
From: <sip:[email protected]>;tag=000000000051032-A7856F6B_lnqxjfl-234D57BF
To: <sip:[email protected]>

3458152 is my sipnet.ru account.
dsenash4 is CallerID for incoming trunk.

I reroute the call to outgoing SIP trunk to IVR.

Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=972dfe12

257 is number of IVR.
231 is AuthID which I had to put when I created this trunk because it is mandatory. Actually IVR doesn't need authentication for incoming calls.

I would like to see incoming CallerID dsenash4 instead 231.

 

[archie]

I have experience in telecommunication more that 12 years. I know a lot of switches. And only some worst switches can't do this.
I don't believe that it is hard to do so.

As I already told you, it is easily achieved if IVR registers as an extension. Our IVR does it. If yours doesn't - I don't think it is our problem. Adapt your IVR to work as extension than.

This is important for two reasons:
- It´s nice to be able to show the callers real name when making direct SIP calls and the callers phone number when making calls to PSTN.
- We love the feature "Append Queue Name to Caller ID" but can't use it with our current settings. When somebode makes a call that go through a queue or ring group and the call is routed out through the trunk, the queue name gets appended to the From: User part of Invite which makes the provider refuse the call due to "Illegal A-number" (we need to place a valid phone number without anything appended in the user part).

I will think about it. Probably we will introduce additional variables for Outbound Parameters.

 

[[email protected]]

Thanks!

 

[SY]

Stepan,
Example:
I have incoming INVITE.

Request-Line: INVITE sip:[email protected]:18973;rinstance=c4786e7da226d379 SIP/2.0
From: <sip:[email protected]>;tag=000000000051032-A7856F6B_lnqxjfl-234D57BF
To: <sip:[email protected]>

3458152 is my sipnet.ru account.
dsenash4 is CallerID for incoming trunk.

I reroute the call to outgoing SIP trunk to IVR.

Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=972dfe12

257 is number of IVR.
231 is AuthID which I had to put when I created this trunk because it is mandatory. Actually IVR doesn't need authentication for incoming calls.

I would like to see incoming CallerID dsenash4 instead 231.

You didn't answer the question 2:

2. "I need to generate following INVITE to my VoIP/PSTN trunk" (sample which is based on 1.)

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

I see I touched very important issue.

It will be clear after the answer to "question 2" will be provided

Thanks

 

[dsen]

Archie.
It is discussion like "Who was first. Egg or hen." Like this who is first? Seller or buyer. And next what need to do first: To develop or to sell? I believe "to develop".
Any Telecom switch without interaction with other equipment is nothing. You must to try to connection with other equipment because it is telecommunication eqipment. it is it's main purpose. To communicate.

I didn't create my IVR. And I can't change it. Your IVR will not replace it. Because my IVR works over WebServices and VXML with internet. Your IVR doesn't support this.

The root reason of my request was suggestion that it is th bug and can be fixed easily.
If not - Ok. Tell me: We will not fix this. It is enough answer for me as well. I will then tell my partners: "3CX will not support this. Let's buy something other".

The answer "Not" is acceptable for me as well. I am not happy about this. I like 3CX but I have to humble.

 

[dsen]

Stepan,
I answered.

Your question:
2. "I need to generate following INVITE to my VoIP/PSTN trunk" (sample which is based on 1.)

My answer above: I would like to see incoming CallerID dsenash4 instead 231.

Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Contact: <sip:[email protected]:5060> <--- there
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=972dfe12 <---- there

Please read my post above. I did full description above.

 

[archie]

It is discussion like "Who was first. Egg or hen."

I'm not a philosopher. I'm not even a seller. I'm developer. So let me skip this part.

You must to try to connection with other equipment because it is telecommunication eqipment.

And we do so. But it is clear we can not handle everything that exists. If someone ask me to support H.323 because it is also telecommunication protocol - I will refuse by the same reason.

Because my IVR works over WebServices and VXML with internet. Your IVR doesn't support this.

Our IVR does support this. But it is not trivial and requires some (though not much) programming skills. That's why we do not advertise such ability widely.

 

[SY]

Stepan,
I answered.

Your question:
2. "I need to generate following INVITE to my VoIP/PSTN trunk" (sample which is based on 1.)

My answer above: I would like to see incoming CallerID dsenash4 instead 231.

Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Contact: <sip:[email protected]:5060> <--- there
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=972dfe12 <---- there

Please read my post above. I did full description above.

Ok,
Could you please provide detailed description of entity "231"?
Why does "231" appear in case if call comes from "[email protected]", (originally) addressed to "[email protected]" and should be delivered to "[email protected]"?
What is the role of "231"?

Thanks