I think you can do the following: make the FXO device as a 2 stage dialing normally, and you can forward the calls to it as the following, from the ARS: - for example if you need to dial 9 and then dial any local number so you can make the ARS as the following: ARS: patton- OUT Matching Pattern: To ...

Dear All we need to improve the normal call waiting for our service which based on Brekeke PBX. Now, we are using a Brekeke PBX, but we did not configure any users on the PBX to eleminate it's license users limitations. we connect our PBX to SVK Billing System for AAA functions. Now, our UA ATA ...

yuyuyu

please try to make sure that your GW have the ability to be Trunking GW, to be able to send any calls comes to him to the PSTN directly. to check that try to call the account of the GW, and after hearing the PSTN Tone, dial your home number. if this happened, so your GW not in the Trunk GW mode ...

Now I use the PBX 2.1 Beta Version as an evaluation version.

please, I need a solution for this problem:

http://www.brekeke-sip.com/bbs/viewtopic.php?t=5264

Thank you very much

Wht are the contact URI and requester? the contact URI : sip:33@192.168.1.30:5060 which is the IP of the PBX B, the Requester: Requester : 62.150.107.168:1160 which is the IP of the ADSL router which the PBX B is connected to the internet through it. I put the network interface: 192.168.1.1, 62.150 ...

Dear Peng and Nick Now I'm facing another problem, When I want to call from the upper PBX ( PBX A) to PBX B( which registered to PBX A, U found that the call does not go through to the PBX B, and I cant hear the greating of the PBX B. Note : before the last modification in the Dial plan, this call ...

Dear Peng Now I make it as the following: Matching pattern: $port=15060 $localhost=true $request=^REGISTER Deploy pattern: $action=register $auth=false &net.registrar.onlyglobal=false and its working too much fine. now I can call the PBX A Extensions, and also I can call any Extension. Thank you for ...

Dear Peng

what you want to do with this dial plan?

I make it "&net.registrar.thru.catchhere=true", but no change in the action, the same problem appears.

Do you have any idea about the suitable dial plan for this situation.

Thank you
Best Regards

Dear Nick I'm using PBX 2.0.7.2 on windows 2003 server. Now I was make an upgrade for the pbx with the 2.1 Beta version, but the same problem appears. Do you try before to connect Brekeke PBX to an ITSP, and in the same time connect it to a PSTN GW to keep the extensions in touch with the local ...

I think you can make an authentication for the user in case of registration and invitation, from sip server configuration --> Sip( General).

Thank you
Amr

hi pablo The problem I think does not in ring time( I was cange it as you told but not solve the problem), but I think its from that the PBX B registered to PBX A, and not registered to itself Sip server, So that there is no call routing between the PBX B and its Sip server. Do you think that this ...

Dear All - I have main Brekeke PBX, and I need to make another PBX which will be registered to the main PBX sip server. - I can call any UA which connect to the main PBX from the internal PBX. - when I call from UA( which connected to main PBX), to the internal PBX, and want to call any internal ...

Dear All

I'd like to ask about that, does the Brekeke support sip and rtp encryption, so that we can encrypt our sip and rtp packets in some countries, to overcome the blocking of the sip traffic.


thank yuo

Dear Sir

it's already like that, but still some users can call in one direction only( can call out, but can not receive calls), so there is any additional configuration for this situation.

Thank you

Dear Peng

can you send me the suitable dial plan for this situation??

Thank you
Amr

Dear Nick

My Sip server not behind firewall, I need the system to support firewall traversal to be able for the clients all over the world to call through my Server in two ways.

Thank you
Amr

Dear all

Do any one can give an Idea about how to make our SIP Server to be firewall traversal , because some of our users can send but not receive the calls.

How we can solve this problem?

Thank you
Best Regards
Amr

all soft and hard phones do the same.

but when I make multidomain = no in the sip server configuration, the two domains can reach each other and works as a one system.

Thank you for your support to me.

we are using :

- Linksys PAP2T (ATA Box)
- Koncept KTA2110
- Duxoft soft phone


Does it's related to the UA device.

Dear All We have a problem, and we can describe it as follows: 1- we have an OnDo PBX. 2- we have a two domain names : sip.a.com, and sip.b.com, which directed to the same IP address. 3- some users registered to the system with sip.a.com , and other registered with sip.b.com . 3- any user with sip.a ...

Dear Support Team we have a problem with OnDo PBX, we can discribe it as the following: 1- we are using an SVK software billing system. 2- when we call any international number, the call start and the accouting start normally. 3- in case of internet connection failure from the caller side, the call ...

1. Brekeke Product Name and version: 2. Java version: 3. OS type and the version: 4. UA (phone), gateway or other hardware/software involved: 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : 6. Your problem: I have a situation, and need to discuss it ...

I face the same problem, but with the higher number of users, some of my users working fine, but some other not ringing but registering ok and can call from it normally.