CID doesn't pass to provider

I use providers that all support setting CID or passing incoming CID to PSTN: callwithus.com, voip.ms, etc.

On calls from PSTN to another PSTN via PBXes trunk, I noticed that original CID in form “name” <number> gets replaced by “name” <user@provider>. Provider doesn’t get the original PSTN number part of CID, so it sends junk or random numbers to PSTN destination.

Please note I’m talking about passing CID of incoming call – it’s different every time and thus cannot be set static in trunk.

Example:
Call with CID “My Name” <15142222222> arrives to the provider as “My Name” <user@sip.provider.com>

This information verified with provider, which sent me log showing incoming CID without the original number.

I hope someone can look into that

This post has been edited 1 time(s), it was last edited by dor on 17.07.2008 at 19:46.

RE: CID doesn't pass to provider

Writing the above mentioned $-Codes into the trunk is not static, and did you try leaving “My Name” of your classical extension empty?

Also keep in mind, that if you try to pass the CID Name to a mobile phone, or a fixed line outside N. America, it will most likely fail to appear on the display.

i-p-tel - you were right, with callwithus.com, once I managed to prefix the CID with '1' to comply with e164, CID appeared on my mobile with no problem.

It didn't work with voip.ms though, I asked them what could be wrong there.

Could you tell me one thing - when CID is not in right format it was replaced by <user@provider> - who does it, PBXes or provider?

This post has been edited 1 time(s), it was last edited by dor on 18.07.2008 at 01:01.

I use voicetrading and cat set the cid I want.

I also pass incoming call on to my mobile phone and am able to pass on the incoming number.

For that reason I have created a trunk that calls my mobile or other pstn extension.

Normally you would fill in your username but leave this on blank
in password field fill in Password:username
dont forget the :

Now setup a pstn extension and use the new trunk. don't fill in any cid numbers here.

Using voicetrading the caller id will be passed to my mobile number.

RE: CID doesn't pass to provider

Zitat:

Originally posted by i-p-tel Writing the above mentioned $-Codes into the trunk is not static, and did you try leaving “My Name” of your classical extension empty?

Guys, there’s definitely a problem here. Since putting <1${DOLLAR}{CALLERIDNUM}> into trunk’s outbound CID resolved the problem only partially, I went to provider again for some more extensive testing.

Here’s what’s happening, and please note – incoming CID from DID provider is always looks as “Doronin Test” <5141111111>, 10 digits.

If I don’t set trunk’s outbound CID at all, provider receives the following: name and URI, as you can see number 5141111111 isn’t sent there at all.
From: "Doronin Test" <sip:188778888@sip.callwithus.com>

If I set trunk’s outbound CID to <${DOLLAR}{CALLERIDNUM}>, with no any prefix, it eliminates the name part from CID, and provider receives the following:
From: "5141111111" <sip: 188778888@sip.callwithus.com>
As you can see number 5141111111 goes instead of the name, and again URI sent instead of the number. In this case provider picks CID from name field, and this is why it appears to work sometimes. It has nothing to do with e164

To confirm the above, I set trunk’s outbound CID to "Doronin Test" <${DOLLAR}{CALLERIDNUM}>
As expected, provider received again only name part and URI
From: "Doronin Test" <sip:188778888@sip.callwithus.com>


Guys, all this caused by the fact that PBXes always send URI like <sip:188778888@sip.callwithus.com> instead of the number. Now I can understand all weird results I see in other providers call monitor lists. It would be much helpful if you fix it to send CID in its original form.

Thanks!!

RE: CID doesn't pass to provider

That's the SIP standard. The From field takes your account URI at the provider. The name part is left for CIDs.

Zitat:

Originally posted by bobmats Normally you would fill in your username but leave this on blank in password field fill in Password:username dont forget the :

bobmats - your workaround worked for ALL providers I have tried!

Thanks!

P.S. Just received feedback from provider. When configured this way PBXes sends everything as expected:

<--- SIP read from 64.118.93.76:28820 --->
INVITE sip:15142222222@sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 64.118.93.76:28820;branch=z9hG4bK03f002bb;rport
From: "Doronin Test" <sip:5149980000@sip.callwithus.com>;tag=as6bdc9759
To: <sip:15142222222@sip.callwithus.com>

P.P.S. All this probably means that the root cause of CID troubles lays in the fact that URI of the FROM field should be in format
PSTN_number@sip.provider.com
and not
user@sip.provider.com

Does it make sense?

This post has been edited 4 time(s), it was last edited by dor on 18.07.2008 at 23:18.

Some background info regarding SIP and the Off-Net realm

We should keep in mind when the SIP protocol was conceived, it was designed to allow ubiquitous communications (voice, video, messaging) between users of broadband networks. SIP was not designed or intended as an ISUP or SS7 replacement, which are the underlying protocols of the PSTN telephone system worldwide.

Through its' versatility, expandability and flexibility the SIP protocol is currently able to handle most of the ISUP call flows natively, almost all of the PSTN features we take for granted today, as well as the advanced functionality it was designed for initially, which should follow us well in the future. Regarding the rest of the PSTN centric and the GSM / 3G centric ISUP messages, the SIP-T and SIP-I standard extensions were introduced, to handle them properly and to allow interoperation with the legacy telephone networks.

So when making bold claims, the SIP URIs should have been this way or the other, a lot of carefully selected choices and details are glossed over, for questionable gains which can be achieved via other means. SIP URIs resemble email addresses, so messages between them can be exchanged using similar methods email messages employ. Phone numbers are notations of the PSTN world, which were carried forward into the SIP realm to make it intuitive for us to use, but they are not essential to communicate as they were in the PSTN realm.

RE: Some background info regarding SIP and the Off-Net realm

Zitat:

Originally posted by DiaforaSo when making bold claims, the SIP URIs should have been this way or the other, a lot of carefully selected choices and details are glossed over, for questionable gains which can be achieved via other means. SIP URIs resemble email addresses, so messages between them can be exchanged using similar methods email messages employ. Phone numbers are notations of the PSTN world, which were carried forward into the SIP realm to make it intuitive for us to use, but they are not essential to communicate as they were in the PSTN realm.

Well, I my bold statements about SIP URI I intended to merely point on discrepancy on how PBXes sends CID data depending on the way user/password set up. As far as workaround works this is not that big deal.
However, as a software developer myself, I'm having hard time to understand why would you make your customers to do this kind of effort to merely find a way to pass CID in a form that is "standard" to "real world" providers.

RE: Change CID

Just a follow-up. I've since been able to forward CID information to my provider Les.net using the 1${DOLLAR}{CALLERNUMID} in outgoing CID setting. However, my other provider Vitelity is still not able to recognize that.

Unfortunately, bobmats' workaround doesn't seem to work for me as these providers either do not allow non-registered trunks to make outgoing calls or do not allow the format of password:username in the password field. If anyone else using these providers know otherwise, please kindly let me know.

RE: Change CID

Leave your registered trunk as is, and create another one, not registered, with the same credentials but in bobmats way.
Send your outgoing calls via that new trunk.

RE: Change CID

the way I do it only works for providers that allow you to set any outgoing id.

for example I use voicetrading which allows me to set any cid I want.

voicetrading also has other options where I have to verify my cid and only the verified cid works like this.

I also have a trunk that calls my cellphone as an extension. The username is left blank in the trunk and I only use passowrd:username in that field.
This way the cid of the caller gets forwarded to my cellphone so I can see who is calling.

RE: Change CID

Zitat:

Originally posted by doronin However, as a software developer myself, I'm having hard time to understand why would you make your customers to do this kind of effort to merely find a way to pass CID in a form that is "standard" to "real world" providers.

Well, we didn't design the SIP protocol or how it interacts with ISUP, the underlying PSTN protocol. We didn't create the various SIP Proxies available, or run the ITSPs who provide us with SIP Trunks to call PSTN numbers.

On the other hand, keep in mind that you can't pass the CallerID you desire with any of the "real world" providers, namely the telcos who use TDM Class-5 switches. When you Call Forward your fixed line at home to your mobile phone, there is no way the "real world" telco will display the CallerID of the original inbound call to your home PSTN line, but the displayed number on your mobile will be your home number.

All this functionality became available with VoIP telephony, precisely because the telephone numbers are no longer essential to the underlying process for communication between users of the network.

RE: Change CID

doronin, bobmats:

The method doesn't seem to work with Vitelity. I have a trunk registered to the server, then another trunk with the same credentials but in password:username format. Any time I use this outgoing trunk I get a circuit-busy from the log.

Vitelity does allow me to set CallerID, but it's on their portal and is a static number.

I tried troubleshooting this with Vitelity, but after a couple tries I got this in response:

Zitat:

You are not setting it properly. At this time we can only suggest that you check with your PBX vendor and/or administrator for further assistance. We do not provide this type of technical assistance.

RE: Change CID

Zitat:

Originally posted by bobmats Just tried this with voicetrading to forward the caller id to my mobile and it works. I have setup a trunk for outgoing calls using voicetrading. username: leave empty password: password:voicetradingusername than I created an outboud route which uses the above trunk. Under dial pattern you for example add 33|xx. under extension add pstn extension leave caller id empty under pstnnumber type: 3300.... (your number) When someone now calls me on a did number using caller id pbxes will forward the call to the pstn extension (if you have set the extension in the ring group for that caller id) and the caller id number will get forwarded to my mobile (pstn extension)

You're my hero! I've spent ages digging around trying to find how I can force the CID for voicecheap!

I did as you suggested however under "username" I put my authorised phone number in the 0044 format then set the password to be password:username

All works well ))))

RE: Change CID

I communicated with Vitelity about what they need in order to set the outbound caller ID. Here is what they said.

Zitat:

Our system reads the caller ID set by your equipment from within the From field in the initial SIP invite. From: \"Vitelity\" <sip:7202575400@64.2.142.25>;tag=as517db3b1

I've tried the bobmats technique of leaving the username blank and putting the password and username in the password field. But that did not work.

Does anyone know if PBXes can be configured to send the outbond caller ID for calls forwarded to my cell phone in the way Vitelity likes?

Thanks in advance!

RE: Change CID

I am also having trouble having the CID I have entered on either the extension or trunk screens passed through to my DID provider, Callcentric. The correct outbound CID is showing up in the PBXes.com call monitor but is not getting to Callcentric.

They say that sendrpid must be set to yes and trutrpid must be set to no in the Asterisk sip.conf file.

Looking at the PBXes.org source view for sip.conf., neither variable appears even when I have placed an phone number entry for outgoing caller id in the trunk or extension screen.

Could this be the problem? If so, could you had a setting to the UI to allow these variables to be optionally set?

Thanks.

RE: Change CID

The Flowroute FAQ states:

Zitat:

We transmit Caller-ID based on the presence of one of the following header fields in order of preference: "P-Asserted-Identity", "Remote-Party-ID" or "From:".

Before I purchase a block of time from Flowroute, can anyone confirm that I can configure my PBXES account to:

• set the CID to my choice when I initiate a call
  
• maintain the call originator's CID when an incoming call is routed to my cell phone via Flowroute
  

RE: Change CID

The same FAQ page at Flowrate states: "We'll give you $0.25 free test credit when you signup."

How about sending a couple of calls without answering them, to verify what happens with CID delivery on both cases.

Have nice day.