Hi Hope and Harold, This dial plan works within our environment but not with our ITSP,so I guess our ITSP is not sending the CLI to the end point.I'll check with them and let you know,thanks. Matching Patterns $request = ^INVITE To = sip:11(.+)@ From = sip:(.+)@ Deploy Patterns $auth = false To ...

Matching Patterns $request = ^INVITE From = sip:11(.+)@ To = sip:(.+)@ Deploy Patterns $auth = false From = sip:+1%1@ To = sip:%2@ITSP ip With the above dial plan I get error 603.The reason is because +1 is placed in front of the dialed number. However if I switch the To and From position like this ...

Hope: Do you need +1 added to caller number or destination number? Vincent: I need +1 to be added to Caller number. Example: if a user registered at Brekeke as 112223333 and this user dial to 4445555 Brekeke will send call with From number +12223333 and To number 4445555 is this what you need? Yes ...

Hope: Matching Patterns $request = ^INVITE From = sip:11(.+)@ To = sip:(.+)@ Deploy Patterns $auth = false From = sip:+1%1@ To = sip:%2@ITSP ip with the above dial plan, if a user registered at brekeke such as 112223333 and this user dial to 4445555 brekeke will send call with From number +12223333 ...

Hi Hope and JanP,

I guess the User ID is appended in the "From" header. If that's true,how about this dial plan?

[Matching Pattern]
$Request = ^INVITE
From:sip:11(.+)@
To:sip:(.+)@

[Deploy pattern]
$auth = false
From:sip:%1@
To:sip:+1%2@ITSP ip
continue = false

Thanks.

JanP: I'm trying to pass CLI (Caller ID) to our ITSP. The Caller ID in this case is the registered user account number. Right now we're successfully passing 10 digit CLI but our clients are requesting for 11 digits and our account factory (system) can only generate a maximum of 10 digit account ...

Hope: I want to send the call through the SIP server.What I'm trying to do is to pass CLI (Caller ID) to our ITSP. The Caller ID in this case is the registered user account number. Right now we're successfully passing 10 digit CLI but our clients are requesting for 11 digits and our account factory ...

1. Brekeke Product Name and Version:Bpbx 2.4.9.0 2. Java version:5.1 3. OS type and the version:Window 2008 Server 4. UA (phone), gateway or other hardware/software involved:Cisco,Grandstream and Yealink IP Phone 5. Your problem: I'd like to add prefix 1 to a registered user ID before sending the ...

Thanks Hope,I'll try it and let you know if it works.

is it like caller make a call to number 5347765098 and then hangup and pbx should send call to both caller and 84927788546? Yes that's how it is but caller is actually 84927788546 calling from a cellphone (PSTN) via a DID access number:5347765098. Here's the call flow sequence: (A) 84927788546 a ...

1. Brekeke Product Name and version: Bundled Bpbx 3.0.6.3 pro 2. Java version: 6 3. OS type and the version:Windows server 2008 4. UA (phone), gateway or other hardware/software involved:RadiusCat Billing server. 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...

Try this dial plan,that's what I use: [Matching Patterns] $request = ^INVITE To = sip:0(.+)@ [Deploy Patterns] $auth = false To = sip:%1@ &register.contact.remote = true $continue = false. Remember to add the prefix "0" while sending calls to registered users.For example if a user is register with ...

I've tried the environment variable prefix but it didn't help.Could this be a policy issue?

Hello Ambrosio,

Sorry for the typo,I actually did use the environment variable character "&" in this line of statement :
[&net.rtp.follow.remoteaddr=true] but did otherwise in

"$register.contact.remote=true ". I'll try it and see if it works.

Thanks.

1. Brekeke Product Name and version: BPBX 2.4.9.0 2. Java version:1.6 3. OS type and the version:Win 2008 Server. 4. UA (phone), gateway or other hardware/software involved:GrandStream GXP 280 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : 6. Your ...

Hi Hope,

Sorry for my late response,I've been pretty busy lately. As to your question,the only dial plans I have are the ones above.

Any more ideas on what I'm missing out?

Hi Hope,thanks for your response.I created a pbx user for Grandstream,same user name and password as in the the Grandstream and Sip server but it didn't work,I'm getting error 404 not found instead. Must I have a pbx user defined to send calls to an ITSP? since the pbx user can not be found,what do ...

1. Brekeke Product Name and version:Bpbx 2. Java version:2.4.7.3 3. OS type and the version:win 2003 standard 4. UA (phone), gateway or other hardware/software involved:Grandstream GXP280, Asterisk Gateway 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns ...

"Brekeke Sip UA" in this context is our Callback Module and "12358183385005 is registered in our SIP Server.

Hi JanP,

Thanks for your response.

To answer your question,yes "55060" is our Sip hostname.

Thanks again,
Vincent.

1. Brekeke Product Name and version:Sip Server / Pbx 2. Java version:2.1.6.6 pro 3. OS type and the version:Win 2003 standard edition 4. UA (phone), gateway or other hardware/software involved:AAA RadiusCat,IVRcat,Callback ,Calling Card,ANI Callback and Web Callback modules. 5. Select your network ...

Haddas,

Thanks for your response,I'll contact Brekeke team as suggested.

Thanks again,
Vin.

Haddas,

Thanks for your response.We actually have JTAPI running in our server.Without JTAPI our IVR wouldn't work.

Since it's possible to create a dial plan that can meet our need.Could you please show us a sample dial plan?

Thanks for your time and kind assistance.

Regards,
Vincent.

1. Brekeke Product Name and version:Sip/pbx version 2.1.6.6 basic 2. Java version:1.5.0_16 3. OS type and the version:2003 standard edition 4. UA (phone), gateway or other hardware/software involved:RadiusCat AAA,Quintum gateway 5. Select your network pattern from http://www.brekeke-sip.com/bbs ...

1. Brekeke Product Name and version:pbx version 2.1.6.6 Basic 2. Java version: 3. OS type and the version:win 2003 standard edition 4. UA (phone), gateway or other hardware/software involved:radiusCat AAA,Quintum gateway,Brekeke sip server 5. Select your network pattern from http://www.brekeke-sip ...