Search found 13 matches
- Tue Sep 07, 2010 5:52 am
- Forum: Brekeke SIP Server Forum
- Topic: Nortel 2-Way audio issues with Rauland
- Replies: 19
- Views: 22184
ports 10000-10999 are available on the server. in addition, i forgot to mention in my initial post, i am able to successfully connect to my room's endpoints every time with 2-way audio if i use x-lite connected to the BBS. mem/cpu is not high we making making calls. the 50 calls are performed one ...
- Fri Sep 03, 2010 10:15 am
- Forum: Brekeke SIP Server Forum
- Topic: Nortel 2-Way audio issues with Rauland
- Replies: 19
- Views: 22184
The &net.sip.replacesdp.multipart as posted is the copy/paste from brekeke. I am not able to attain the packets from the handset side currently. Client is not behind a NAT. any thoughts of why a call may work fine with 2-way audio every 20 calls or so? the SIP setup never fails, it is only the RTP ...
- Fri Sep 03, 2010 6:59 am
- Forum: Brekeke SIP Server Forum
- Topic: Nortel 2-Way audio issues with Rauland
- Replies: 19
- Views: 22184
Nortel 2-Way audio issues with Rauland
1. Brekeke Product Name and version:Brekeke SIP Server/2.3.8.2/286 2. Java version:1.6.0_16 3. OS type and the version:Windows 2003 R2 4. UA (phone), gateway or other hardware/software involved: Rauland Responder 5/ Nortel CS1k 5. Select your network pattern from http://www.brekeke-sip.com/bbs ...
- Thu Aug 12, 2010 11:20 am
- Forum: Brekeke SIP Server Forum
- Topic: setting up Brekeke with Rauland R5 and Ascom Phone System
- Replies: 10
- Views: 16725
Brian, I know that ASCOM requires you to setup an account for them to use. I used "9999" for them to register to brekeke. In addition, we had an issue at one point, where ASCOM's calls were not coming from the AA60, but from the VoiP gateway. They way you have your Brekeke designed, it should work ...
- Wed Nov 04, 2009 9:26 am
- Forum: Brekeke SIP Server Forum
- Topic: Complicated Dial Plan Help
- Replies: 5
- Views: 6268
i am able to establish the call based on this dial plan: Matching: $request=^INVITE To=sip:([0-9]{3})([0-9]{4})([0-9]{1,2});.*@ Deploy: To=sip:%1%2*%3@IP Matching: $request=^INVITE To=To=sip:([0-9]{3})([0-9]{4});.*@ Deploy: To=sip:%1%2@IP however, no matter which order i place them the first near ...
- Mon Nov 02, 2009 8:27 am
- Forum: Brekeke SIP Server Forum
- Topic: Complicated Dial Plan Help
- Replies: 5
- Views: 6268
Yes the destination URI's are registered. here is the syntax i got working after working with Support using an X-Lite Phone: Dial Plan 1 Matching: $request=^INVITE To=sip:([0-9]{6})([0-9]{4})@ Deploy To=sip:%1*%2@IP Address Dial Plan 2 matching: $request=^INVITE To=sip:([0-9]{3})([0-9]{4})([0-9]{1,2 ...
- Fri Oct 30, 2009 10:29 am
- Forum: Brekeke SIP Server Forum
- Topic: Cisco CallManager 6 to SIP Server via SIP Trunk
- Replies: 9
- Views: 15721
Justin to direct your calls for the emergency calls in the Rauland system you would have to have "Teams" built in the Responder5 system. For example: Code Blue Team This team would be set to only receive calls from code blue buttons. Then in the Responder5 application, the members of that team would ...
- Fri Oct 30, 2009 9:10 am
- Forum: Brekeke SIP Server Forum
- Topic: Complicated Dial Plan Help
- Replies: 5
- Views: 6268
Complicated Dial Plan Help
1. Brekeke Product Name and version:Brekeke SIP Server 2.3.8.2 2. Java version:1.6.0_16 3. OS type and the version: Windows 2003 R2 Standard 4. UA (phone), gateway or other hardware/software involved: Nortel CS1000E 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...
- Fri Jul 17, 2009 5:17 am
- Forum: Brekeke SIP Server Forum
- Topic: How to Specifify a ringtone based on Caller ID FROM
- Replies: 11
- Views: 13912
How to Specifify a ringtone based on Caller ID FROM
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 would like to set a ring tone on a cisco ...
- Thu Jul 16, 2009 6:12 pm
- Forum: Brekeke SIP Server Forum
- Topic: strip leading characters from caller id
- Replies: 11
- Views: 9594
got mine to work. It was a port conflict between the Rauland software and Brekeke. I was able to change the port on the Rauland system and it worked fine. Also, i had to force RTP = ON as well to get the Voice Paths to work. THanks for your help. Also, is there a guide with the different variables ...
- Thu Jul 16, 2009 6:07 pm
- Forum: Brekeke SIP Server Forum
- Topic: Cisco CallManager 6 to SIP Server via SIP Trunk
- Replies: 9
- Views: 15721
I was able to get this working Justin. I have the Brekeke actually loaded on the Rauland Gateway server, which in turn uses port5060. so i had a conflict there. So i had to change the Responder Gateway Server to use a different port (i used 5061). In addition, i had to set the RTP to "ON" rather ...
- Mon Jul 13, 2009 10:16 am
- Forum: Brekeke SIP Server Forum
- Topic: Cisco CallManager 6 to SIP Server via SIP Trunk
- Replies: 9
- Views: 15721
i am interested in this as well. Since i am doing the same thing as you Justin. I have Brekeke loaded onto the RGS server (got it working finally by changing the SIP port RGS uses) i have a dial plan created Matching Patterns: $request= ^INVITE Deploy: $target = IP Address of SIP listener provided ...
- Fri Jul 10, 2009 6:44 am
- Forum: Brekeke SIP Server Forum
- Topic: strip leading characters from caller id
- Replies: 11
- Views: 9594