transit calls

[Nikolay Kondratyev]

Does PBXnSIP support transit calls?

If yes, how do i setup it?

If not, is it planned for future development?

 

Thanks in advance,

Nikolay.

[Vodia PBX]

Does PBXnSIP support transit calls?

If yes, how do i setup it?

If not, is it planned for future development?

 

Can you elaborate on what a transit call is?

[Nikolay Kondratyev]

Can you elaborate on what a transit call is?

 

I mean the following: get the call from a sip trunk and depending on the number dialed, terminate it on a local extension or route it to another sip trunk.

 

The thing is our company is going to be pbxnsip reseller in Russia and we have several potential clients for distributed ip telephony systems.

And we are evaluating where we can offer pbxnsip.

The typical scenario could be:

Meidum corporate (couple hundreds of subscribers). Main office and several branch offices with pbxnsip in each office. Common PSTN gateway(s) or ITSP link(s).

The incoming call must be routed to appropriate office according to telephone numer.

Incoming (from PSTN) calls should be routed on main office pbx, i mean that routing table on a gateway (and espesially on ITSP equipment) is unwanted.

I understand that one could have just one pbx in main office and create domains for branch offices, but this is not always suitable mainly because of the ip link failure paossibility.

That's why i'm asking about transit calls possibility..

 

Thanks in advance,

Nikolay.

[shopcomputer]

I mean the following: get the call from a sip trunk and depending on the number dialed, terminate it on a local extension or route it to another sip trunk.

 

The thing is our company is going to be pbxnsip reseller in Russia and we have several potential clients for distributed ip telephony systems.

And we are evaluating where we can offer pbxnsip.

The typical scenario could be:

Meidum corporate (couple hundreds of subscribers). Main office and several branch offices with pbxnsip in each office. Common PSTN gateway(s) or ITSP link(s).

The incoming call must be routed to appropriate office according to telephone numer.

Incoming (from PSTN) calls should be routed on main office pbx, i mean that routing table on a gateway (and espesially on ITSP equipment) is unwanted.

I understand that one could have just one pbx in main office and create domains for branch offices, but this is not always suitable mainly because of the ip link failure paossibility.

That's why i'm asking about transit calls possibility..

 

Thanks in advance,

Nikolay.

You can have trunks between multiple PBXnSIP servers at various locations, and have the PBX send the extension numbers for the other offices over the trunks.

Set Office 1 with 3xx extensions office 2 4xx entensions, etc. and set the dial plan to use the corresponding trunk.

[Nikolay Kondratyev]

You can have trunks between multiple PBXnSIP servers at various locations, and have the PBX send the extension numbers for the other offices over the trunks.

Set Office 1 with 3xx extensions office 2 4xx entensions, etc. and set the dial plan to use the corresponding trunk.

Thanks for teh reply.

But dial plan is used for outgoing calls (i.e. originated from a registered extension).

Incoming call does not even go through any dialplan. But i need to terminate incoming call on a local extension, or route it to another trunk, depending on the dialed number.

The only exception, as far as i understand, when incoming call goes through dial plan, is when "Assume that call comes from user" field in the trunk configuration is used.

 

I tried to use this field. And i was able to do what i want, but i found a problem.

Let me describe:

I created a special extension 9999, with a special dialplan.

Put 9999 into the "Assume that call comes from user" field of a trunk named "m600".

The dialplan for ext. 9999 is named "incoming_dialplan" and looks as following:

pref trunk pattern replacement

100 "call extension" ^633([0-9]{4})@.* \1

120 m600 ^632([0-9]{4})@.* sip:\1@\r

 

When the call arrives from trunk "m600" it is sent into "incoming_dialplan" (i can see it in the log).

When the dialed number is 632xxxx the call is successfully routed back into the "m600" trunk.

When the dialed number is 6333801 the call is successfully routed to existing and registered extension 3801.

So ... it looks like i got what i wanted, but...

When the dialed number is 6333808 (3808 is configured, but not registered extension) the calling party hears "comfort noise" instead of voicemail prompt.

When 3801 calls 3808 the voicemail prompt is ok.

When "Assume that call comes from user" field is empty and "Send call to extension" field is used, and external caller dials 6333808, he can hear voicemail prompt.

 

Could it be configuration problem?

Or is it a software problem?

 

Should i send any additional info? Please advise what should i send, log file or trace, or something else...

I use 3.1.0.3043 (Centos) version.

 

Thanks in advance,

Nikolay.

[Vodia PBX]

Hmm. If the caller calls a resource on the PBX (say an extension, or an auto attendant, or something else), the PBX is able to involve outbound calls. For example, when calling an extension, the PBX can fork the call also to a phone number. Or when calling an auto attendant, the call can also be redirected to an external number. So far, that seems to make most of the users happy.

 

The only "problem" we had so far was Microsoft Exchange and it's click to dial feature. Because then the call comes in on the Exchange trunk and it is supposed to go out on another trunk. That problem we did solve with the "Accept Redirect" flag and the "Assume call comes from extension" setting. In this case, all calls from that trunk are redirected to an outgoing number - even if the call might go to a local resource.

 

I believe your problem can be solved just by using the "redirect all" feature of the PBX. Just use a local extension and then redirect all calls to the PSTN number.

[Nikolay Kondratyev]

Hmm. If the caller calls a resource on the PBX (say an extension, or an auto attendant, or something else), the PBX is able to involve outbound calls. For example, when calling an extension, the PBX can fork the call also to a phone number. Or when calling an auto attendant, the call can also be redirected to an external number. So far, that seems to make most of the users happy.

 

The only "problem" we had so far was Microsoft Exchange and it's click to dial feature. Because then the call comes in on the Exchange trunk and it is supposed to go out on another trunk. That problem we did solve with the "Accept Redirect" flag and the "Assume call comes from extension" setting. In this case, all calls from that trunk are redirected to an outgoing number - even if the call might go to a local resource.

 

I believe your problem can be solved just by using the "redirect all" feature of the PBX. Just use a local extension and then redirect all calls to the PSTN number.

 

Do you mean, that i should redirect all calls to the same number?

Or do you mean that i should use separate extension for each destination number, that is not a local extension? (It means that customer will have to pay for each destination number, which is being routed to different sip server.)

What i'm talking about is quite standard task: routing of incoming calls between several offices...

Routing is making a decision (depending on dialed number) if incoming call should be sent to local extension (phone, vm, aa, acd queue... any type of extension) or it should be sent to another sip trunk (trunk selection should also be possible based on dialed number).

 

Many other working features of pbxnsip are quite greate, but...Without transit calls routing pbxnsip just does not suite for customers with several pbxes.

 

 

So i have to ask again: can pbxnsip route transit calls?

If not, may be it's worth planning this feature for the nearest development?

 

Thanks and regards,

Nikolay.

[Vodia PBX]

Do you mean, that i should redirect all calls to the same number?

Or do you mean that i should use separate extension for each destination number, that is not a local extension? (It means that customer will have to pay for each destination number, which is being routed to different sip server.)

 

Yes, the seperate extension will solve the problem. Yes, this will require more licenses (think about the margin that you make, hehe). If they are buying another PBX license for the branch office, maybe it is possible to make a package deal, so that the costs are not exploding.

 

What i'm talking about is quite standard task: routing of incoming calls between several offices...

Routing is making a decision (depending on dialed number) if incoming call should be sent to local extension (phone, vm, aa, acd queue... any type of extension) or it should be sent to another sip trunk (trunk selection should also be possible based on dialed number).

 

Without transit calls routing pbxnsip just does not suite for customers with several pbxes.

 

So i have to ask again: can pbxnsip route transit calls?

If not, may be it's worth planning this feature for the nearest development?

 

Today I would technically solve the problem with just more extensions and do a deal with the licenses.

 

We have to check how much side effects it would be if we allow the trunk destination to be either an account or an external number.

[Nikolay Kondratyev]

Yes, the seperate extension will solve the problem. Yes, this will require more licenses (think about the margin that you make, hehe). If they are buying another PBX license for the branch office, maybe it is possible to make a package deal, so that the costs are not exploding.

 

 

 

Today I would technically solve the problem with just more extensions and do a deal with the licenses.

 

We have to check how much side effects it would be if we allow the trunk destination to be either an account or an external number.

Please investigate this task/problem.

First, i was quite happy when i found that regular expressions can be used in the dialplan.

The second time i was happy was when i found that i can assign separate dialplan to each trunk for incoming call analisys (using "Assume that call comes from user" field).

And i was disappointed, when i found that it does not work (i described the problem earlier in this topic).

 

Anyway i'm sure, that after a some time you (i mean pbxnsip team) will arrive at the same idea: tandem/transit routing is a must (if you are going to play at medium corporate market...).

 

By the way, one of the side effect could be as following: P-Asserted Identity, P-Prefered Identity etc. header is unwanted for such tandem calls...

(I mean that pbx should not insert it's own PAI header in such calls).

If i'm allowed to tell my point of view, i would say, that there should be a possibility to select dial plan for incoming calls directly from the trunk configuration page, without reference to an extension. Though, i don't know if it is possible...

 

Thanks and regards,

Nikolay.

[Nikolay Kondratyev]

Another crazy idea:

possibly it'll be more economical to develop a separate product - "pbxnsip-proxy"...

It can be statefull proxy (not b2bua). And will serve just for tandem calls, consolidating several pbxnsip servers into single network...