1. Brekeke Product Name and Version: Brekeke SIP Server 3.8.6.4 2. Java version: Java SE 8 3. OS type and the version: Win2008R2 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: I am using third party database for users database,The selection of Encrypt Users Passwords is ...

Thank for your reply. I found [Configuration]-[Network]-[Interface address 1,2,3,4,5] in the system,What's the use of these settings?Can I fill the three IP address in it,And add route on windows like below: route add ITSP1's IP mask 255.255.255.252 180.136.225.1 route add ITSP2's IP mask 255.255 ...

1. Brekeke Product Name and Version: Version 3.2.4.3 Standard Version 2.4.7.3 Standard 2. Java version:6 3. OS type and the version:win2003 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: If I have three network card on the server,and they link to different network,they ...

janP wrote:> The codec error message is sent from gateway.

It will be the gateway's issue.
Contact the gateway's manufacturer.

Thanks for your reply ,I will contact the manufacturer.

janP wrote:Which gateway product is it?

a 8 E1 gateway, ALLYWLL ETG3008

janP wrote:Which gateway product are you using??

If the caller (UAC) or callee (UAS) does't support G729, a call will not be established.

Use WireShark to check who sent an error response such as 500, 617 and 603.

The codec error message is sent from gateway.

> &net.rtp.audio.payloadtype = 18 Do you know what it means? The payload type 18 means G729. Both UA support G729 audio codec? Thank for your reply. I didn't add &net.rtp.audio.payloadtype = 18 at first,But when 617 error appears,I try to add it. Both UA support G729,if not,all calls will fail,But ...

1. Brekeke Product Name and Version: Version 3.2.4.3 Standard Version 2.4.7.3 Standard 2. Java version:6 3. OS type and the version:win2003 4. UA (phone), gateway or other hardware/software involved:yealink 5. Your problem: I have two sip servers, Server A's version is 2.4.7.3,Server B's version is ...

thank you
Now all problems are solved.
Thank you very much!

:D Thanks for your reply. > From = "%1"<%2> Do you have definitions for %1 and %2 in Matching Patterns? I have no definitions for them.It seems doesn't need to define,My old dail plan can work normally.for example: Matching Patterns: $request=^INVITE To=sip:(1[0-9]{10})@ $geturi(From)=(.+) Deploy ...

1. Brekeke Product Name and version: Version 2.4.7.3 Standard 2. Java version: JRE 6 Update 23 3. OS type and the version: win 2003 4. UA (phone), gateway or other hardware/software involved: 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 3 ...

janP wrote:LeeFrederick,

You can learn how to write Regular expression.
http://en.wikipedia.org/wiki/Regular_expression
http://download.oracle.com/javase/6/doc ... ttern.html

thank you,I am new to sip,and I will learn the Regular expression.

hope,thank you very much,it works normally now.

Thank you very much.
and I also want to limit the length of the phone number,for example,when the length of the phone number is eight and the prefix is "9",It matchs the Deploy Patterns.how do I change the Matching Patterns?
thanks

Thanks for you reply. but I am using BSS,not PBX. I mean when the sip phone dails a phone number begins with zero(for example,02188010000) ,the BSS will route the call to an appointed UA.and also send the phone number(02188010000) to it. I have built the dail plan below: Matching Patterns: $request ...

1. Brekeke Product Name and version: Version 2.4.7.3 Standard 2. Java version: JRE 6 Update 23 3. OS type and the version: win 2003 4. UA (phone), gateway or other hardware/software involved: 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 3 ...

which private IP address are you using?? The SIP Server's private IP address is 192.168.1.6. And I disabled one network card recently,Only the public IP address remains.and the trouble hasn't appear recently.in this case,The voice change connects to the SIP Server throug the public IP address. But ...

Are you using Brekeke SIP Server under a virtual environment such as VMware? Was the CPU usage almost 100% when the trouble happened? The SIP Server is not under a virtal environment,The public IP address is directly configured on the netword card. The CPU uasge was 1%~2% when the trouble happened.

Thanks for your reply.

I can make a call between two ATA devices,but I haven't tried when the trouble happen,I will try next time.in most cases,the talking between ATA and voice exchange is normal.

I donn't want to use the regular port because of the safety reason.

I don't have any dialplan rules. There are no concurrent calls when the issue happens.I found out the trouble when I make a call. All the firewalls are closed, and I have changed the sip's port to 29988.and the RTP port is from 10000 to 10999. I also found when I couldn't open the admintool and ...

I can see both public and private IP addresses at the [interface] field of the [Server Status] page,and it hasn't any other ip addresss there. I don't need DialPlan rule.The voice exchange registers in the SIP server as a sip client,The voice exchange is assigned a sip account,and each talking is ...

Thanks for your reply. I am using a static public IP address as the SIP Server's public IP address,and the ip address is configured on then server's network card. But the server has two network cards,and the other network card has a private IP address.Because the SIP Server need to connect with a ...

1. Brekeke Product Name and version:Brekeke SIP Server Version 2.4.7.3 Standard 2. Java version:JRE 6 Update 23 3. OS type and the version:windows server 2003 4. UA (phone), gateway or other hardware/software involved: 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...