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LED status on cell calls

Mads Mortensen

By Mads Mortensen
in Using Cell Phones

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jim@itstod.se Newbie

jim@itstod.se

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Posted

OK, I understand.

 

But please let me know if this is something that you plan to implement in upcoming versions. We have some customer cases that will be implemented in Q3/Q4 this year and we must know if we can offer them snom ONE or if we must go for other solutions for customers that require BLF status for mobile extensions.

 

If you decide to implement this and want someone to test it, let me know and we will do it.

 

Regards,

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jim@itstod.se Newbie

jim@itstod.se

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Hi again,

 

I thought I would give this a new shot now after the summer holidays. Our hosted competitors in Sweden has this functionality and it is highly requested by customers.

 

Using version 2011-4.3.0.5002 (Win64).

 

In this example I am calling 90400 from my mobile phone. The VoIP provider routes the call to our SIP trunk and puts a "C945" prefix in the To section but we want to use Request URI so this does not matter. What I am trying to achieve is that:

  1. The call should be routed back to the VoIP provider and my extension should show as busy on my colleagues phones busy lamp fields (since 0709355940 is the mobile phone number set on my extension).
  2. My extension ANI should be showed to the person I am calling (SIP From field I suppose), not my mobile phone number.

Here is what happens:

[8] 2011/07/27 23:55:15: Incoming call: Request URI sip:90400@46.59.77.70;user=phone, To is "C94590400" <sip:C94590400@sip-corporate.tele2.se:5060;user=phone>

[8] 2011/07/27 23:55:15: Call from a trunk 1

[8] 2011/07/27 23:55:15: Trunk Tele2 SIP Trunk@sip.itstod.se has country code 46, area code 501

[9] 2011/07/27 23:55:15: Incoming: formatted From is = "0709355940" <sip:+46709355940@212.151.144.8:5060;user=phone>

[9] 2011/07/27 23:55:15: Incoming: formatted To is = "C94590400" <sip:C94590400@sip-corporate.tele2.se:5060;user=phone>

[9] 2011/07/27 23:55:15: Incoming: formatted URI is = sip:90400@sip.itstod.se;user=phone

[8] 2011/07/27 23:55:15: To is "C94590400" <sip:C94590400@sip-corporate.tele2.se:5060;user=phone>, user 0, domain 1

[8] 2011/07/27 23:55:15: Send call to extension is not set. Route the call based on global number 90400

[4] 2011/07/27 23:55:15: Domain trunk Tele2 SIP Trunk@sip.itstod.se could not identify user for 90400

 

The PBX is not routing the call out to the VoIP provider, why?

 

Regards,

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jim@itstod.se Newbie

jim@itstod.se

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Please get back regarding this. If it is not going to happen we must look for another solution for customers asking for this functionality.

Also the reason I need an answer now is that the mobile carrier has applied this setting on my mobile phone to test with the PBX and as it is now I am not able to make any outbound calls at all (since they route all calls to our SIP trunk...). We are not so busy during the summer so we thought this would be a good time to try new things out.

So I need to know:

  • Is it possible at all to route this call out through the trunk?
  • Do you intend to implement such functionality and if so when?

Thanks,

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Kalle Newbie

Kalle

Posted
Posted

Hi again,

 

I thought I would give this a new shot now after the summer holidays. Our hosted competitors in Sweden has this functionality and it is highly requested by customers.

 

Using version 2011-4.3.0.5002 (Win64).

 

In this example I am calling 90400 from my mobile phone. The VoIP provider routes the call to our SIP trunk and puts a "C945" prefix in the To section but we want to use Request URI so this does not matter. What I am trying to achieve is that:

  1. The call should be routed back to the VoIP provider and my extension should show as busy on my colleagues phones busy lamp fields (since 0709355940 is the mobile phone number set on my extension).
  2. My extension ANI should be showed to the person I am calling (SIP From field I suppose), not my mobile phone number.

Here is what happens:

[8] 2011/07/27 23:55:15: Incoming call: Request URI sip:90400@46.59.77.70;user=phone, To is "C94590400" <sip:C94590400@sip-corporate.tele2.se:5060;user=phone>

[8] 2011/07/27 23:55:15: Call from a trunk 1

[8] 2011/07/27 23:55:15: Trunk Tele2 SIP Trunk@sip.itstod.se has country code 46, area code 501

[9] 2011/07/27 23:55:15: Incoming: formatted From is = "0709355940" <sip:+46709355940@212.151.144.8:5060;user=phone>

[9] 2011/07/27 23:55:15: Incoming: formatted To is = "C94590400" <sip:C94590400@sip-corporate.tele2.se:5060;user=phone>

[9] 2011/07/27 23:55:15: Incoming: formatted URI is = sip:90400@sip.itstod.se;user=phone

[8] 2011/07/27 23:55:15: To is "C94590400" <sip:C94590400@sip-corporate.tele2.se:5060;user=phone>, user 0, domain 1

[8] 2011/07/27 23:55:15: Send call to extension is not set. Route the call based on global number 90400

[4] 2011/07/27 23:55:15: Domain trunk Tele2 SIP Trunk@sip.itstod.se could not identify user for 90400

 

The PBX is not routing the call out to the VoIP provider, why?

 

Regards,

 

Hi!

Is this possible (fixed) today with snomONE?

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Kalle Newbie

Kalle

Posted
Posted

We realized that it is not as 'easy' as we thought. So, it is not added to the software yet :(

 

OK, I have "played" some with at Tele2 MEX cell phone. I get the same result as Jim but this is because Tele2 doesn't accept redirects or other calls with somebody elses from: header. So the diversion tag is important.

 

We have since earlier used a "Customer specific header (Example: X-snom: my header)" on trunk with : "Diversion: <{trunk-ani}@{trunk-host}>;reason=unknown" because if not we get "SIP/2.0 403 Call did not pass A-number check" after invite (If we want to send incoming calls a-number when we want to redirect a call to cell - and that we want!)

 

I think it is the same with Jim's example AND our own problem that when a MEX connected cellphone calls it makes a invite to PBX and then the PBX handles it as a redirect.

To make that get accepted by Tele 2 we need to "Accept Redirect:" in trunk and set our A-number connected to Tele2 account in "Assume that call comes from user:".

BUT then PBX sends extension number 600 instead (because the account is 600 0311234567). And the formatting of the Diversion tag that "Accept Redirect:" uses seems to not be accepted by Tele2. PBX sends: "Diversion: <tel:600>;reason=unconditional;screen=no;privacy=off" Should be: "<0311234567@sip-corporate.tele2.se>;reason=unknown"

 

So it would be nice to in section Routing/Redirection if it was possible to set our own Diversion tag? Then the call should be routed correctly and SnomOne is working with Tele2 MEX. And maybe it also signals on BLF? ;-)

 

Please - if you need any testpilot to make this work: I'm in!

 

Regards // Kalle

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Kalle Newbie

Kalle

Posted
Posted

OK, I have "played" some with at Tele2 MEX cell phone. I get the same result as Jim but this is because Tele2 doesn't accept redirects or other calls with somebody elses from: header. So the diversion tag is important.

 

We have since earlier used a "Customer specific header (Example: X-snom: my header)" on trunk with : "Diversion: <{trunk-ani}@{trunk-host}>;reason=unknown" because if not we get "SIP/2.0 403 Call did not pass A-number check" after invite (If we want to send incoming calls a-number when we want to redirect a call to cell - and that we want!)

 

I think it is the same with Jim's example AND our own problem that when a MEX connected cellphone calls it makes a invite to PBX and then the PBX handles it as a redirect.

To make that get accepted by Tele 2 we need to "Accept Redirect:" in trunk and set our A-number connected to Tele2 account in "Assume that call comes from user:".

BUT then PBX sends extension number 600 instead (because the account is 600 0311234567). And the formatting of the Diversion tag that "Accept Redirect:" uses seems to not be accepted by Tele2. PBX sends: "Diversion: <tel:600>;reason=unconditional;screen=no;privacy=off" Should be: "<0311234567@sip-corporate.tele2.se>;reason=unknown"

 

So it would be nice to in section Routing/Redirection if it was possible to set our own Diversion tag? Then the call should be routed correctly and SnomOne is working with Tele2 MEX. And maybe it also signals on BLF? ;-)

 

Please - if you need any testpilot to make this work: I'm in!

 

Regards // Kalle

 

Some more comments of this or am I way off in my mind?

 

// Kalle

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pbx support Newbie

pbx support

Posted
Posted

Some more comments of this or am I way off in my mind?

 

// Kalle

Did you try setting 0311234567 as the ANI on the extension 600? Your case may be different from Jim's, not sure.

 

What Jim wanted earlier was sort of a 2 stage dialing in 1 step, if I remember correctly. Ex: the access number and the destination number will be passed in the same INVITE and PBX "somehow" accepts the call and dials out the destination number.

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Kalle Newbie

Kalle

Posted
Posted

Did you try setting 0311234567 as the ANI on the extension 600? Your case may be different from Jim's, not sure.

 

What Jim wanted earlier was sort of a 2 stage dialing in 1 step, if I remember correctly. Ex: the access number and the destination number will be passed in the same INVITE and PBX "somehow" accepts the call and dials out the destination number.

 

Yes, I do have 0311234567 as ANI on extension600.

 

// Kalle

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Kalle Newbie

Kalle

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Posted

Could you please post an INVITE that you want the PBX to send out? We can take a look at it and try to setup the trunk.

 

Of course, here are some examples, (Incoming number: 0701123456, 0319999600 is PBX main number)

 

 

This is how it should be (Redirected extension 605 to 03190510) :

 

INVITE sip:03190510@sip-corporate.tele2.se;user=phone SIP/2.0

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-581023b173c2c5c06625917d317d58d8;rport

From: "0701123456" <sip:0701123456@pbxdomain;user=phone>;tag=1875464570

To: "Kalle" <sip:0319999605@pbxdomain;user=phone>

Call-ID: 538c560f@pbx

CSeq: 30914 INVITE

Max-Forwards: 70

Contact: <sip:T2username@x.x.x.x:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

In-Reply-To: 121c9d0b-30877060-5b153ef4-966d@212.151.144.8

P-Charging-Vector: icid-value=;icid-generated-at=x.x.x.x;orig-ioi=pbxdomain

Diversion: <0319999605@sip-corporate.tele2.se>;reason=unconditional;screen=no;privacy=off

Content-Type: application/sdp

Content-Length: 384

 

 

This is how it looks like now (Redirected extension 605 to 031901510) :

 

INVITE sip:03190510@sip-corporate.tele2.se;user=phone SIP/2.0

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-5230ea360fea69aa482013409b20702f;rport

From: "0701123456" <sip:0701123456@pbxdomain;user=phone>;tag=1665200458

To: "Kalle" <sip:0319999605@pbxdomain;user=phone>

Call-ID: 29b484c9@pbx

CSeq: 24105 INVITE

Max-Forwards: 70

Contact: <sip:T2username@x.x.x.x:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

Diversion: <tel:605>;reason=unconditional;screen=no;privacy=off

In-Reply-To: 21b0e110-473abd9c-1c04c0-9210@212.151.144.8

P-Charging-Vector: icid-value=;icid-generated-at=x.x.x.x;orig-ioi=pbxdomain

Content-Type: application/sdp

Content-Length: 388

 

 

Redirected on trunk (MEX, 600 is put as "Assume that call comes from user:", 600 has 0319999600 as ANI, X92A in To is a prefix that Tele2 adds because of MEX service.) :

 

[5] 20120503082027: SIP Tx udp:130.244.190.42:5060:

INVITE sip:03190510@sip-corporate.tele2.se;user=phone SIP/2.0

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-60d3f42be369987eb80c3ab60c68e748;rport

From: "0701123456" <sip:0701123456@212.151.144.8;user=phone>;tag=1033792167

To: "X92A03190510" <sip:X92A03190510@pbxdomain;user=phone>

Call-ID: e7bbc063@pbx

CSeq: 29992 INVITE

Max-Forwards: 70

Contact: <sip:T2username@x.x.x.x:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids

Diversion: <tel:600>;reason=unconditional;screen=no;privacy=off

In-Reply-To: 5c1c7fa9-12e21f60-459e15ee-982d@212.151.144.8

P-Charging-Vector: icid-value=;icid-generated-at=x.x.x.x;orig-ioi=pbxdomain

Proxy-Authorization: Digest realm="sip-corporate.tele2.se",nonce="4fa223e6000010847adfab1f8ba6ec7ed797eb8881d4f73f",response="92f40ba7fa5d749cd3abbc6fa796e4da",username="T2username",uri="sip:03190510@sip-corporate.tele2.se;user=phone",algorithm=MD5

Content-Type: application/sdp

Content-Length: 388

 

 

// Kalle

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pbx support Newbie

pbx support

Posted
Posted

Of course, here are some examples, (Incoming number: 0701123456, 0319999600 is PBX main number)

 

This is how it should be (Redirected extension 605 to 03190510) :

 

 

Let's consider this case. For this to work as you expected, you can make use of the "Trunk->Customer specific header (Example: X-snom: my header")".

  • Set the "Extension->ANI" to 0319999605.
  • Set the "trunk->accept redirect" to "No"
  • Set the "Trunk->Customer specific header (Example: X-snom: my header")" to "Diversion:<{ext-ani}@{domain}>;reason=unconditional;screen=no;privacy=off". Note: if you want the trunk's domain to go out in the INVITE, then you can use {trunk-host} instead of {domain}

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Kalle Newbie

Kalle

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Posted

Let's consider this case. For this to work as you expected, you can make use of the "Trunk->Customer specific header (Example: X-snom: my header")".

  • Set the "Extension->ANI" to 0319999605.
  • Set the "trunk->accept redirect" to "No"
  • Set the "Trunk->Customer specific header (Example: X-snom: my header")" to "Diversion:<{ext-ani}@{domain}>;reason=unconditional;screen=no;privacy=off". Note: if you want the trunk's domain to go out in the INVITE, then you can use {trunk-host} instead of {domain}

Thanks for your suggestion.

But I have already tried that as you can see in a previous post. It does make it ok with calls redirected by extension. But it doesn't solve the problem with a mex extension. A mex extension is an outside extension that makes an invite to pbx so trunk needs to redirect. And to make that work the trunk needs to accept redirect. With the extra "Customer specific header" the invite will then have two Diversion tags.. The "Customer specific header" also adds Diversion tag to every outbound call.

I am not sure that my suggestion will be the best or even work but I think that a possibility to edit the DIversion tag that "Accept Redirect" makes would solve it.

// Kalle

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