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Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090612/5a499965/attachment.htm

...arary and includes of the Fedore Core mysql directories. But my box is now running MySQL 5, and the source directory is at /root/mysql-standard-5.0.16-linux-i686directory. My question is how to compile using this new source code for the header files and the libarary files? Many thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060315/bbe8f68a/attachment.htm

...be able to almost see everything that is going on. I want to thank the person that left that tip in the mailing list. Sorry I forgot who it was as I was searching through the entire archive from the begin. I hope that this will help some people when there isn't any one to help you. Regards, Kengie Ho On 7/21/05, Kenige Ho <kengiepanda@gmail.com> wrote: > Dear All, > > I have a problem with the Marco and the Realtime Extensions in my > extensions.conf. The problem is that when I exit from my Marco, I > should return to my calling context, which is default but the next...

...om www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406 Digium Card OS: Fedore Core 5 Thank you for your input. Regards, Kengie

...ome bug in my sytnax or something in the asterisk program itself? Below is my default context: [default] exten => _X.,1,Macro(stdexten,${EXTEN},${CALLERIDNUM}) ;Realtime Routing from MySQL switch => Realtime/default@extensions [macro-stdexten] exten => s,1,NoOp(Leaving Marco) Regards, Kengie Ho

...ac@XXX.XXX.XX.XXX CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:XXXXXXXX@XXX.XXX.XX.XXX> WWW-Authenticate: Digest realm="asterisk", nonce="4929aec7" Content-Length: 0 Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/1db20f5a/attachment.htm

...anged up and tried X-lite again, it was not working. The second call was only a few seconds apart. Moving back to the online dialer, it wasn't working either. So it is just very strange to me how this happened and i was think maybe it was the RTP negiotation. Do you have any ideas? Regards, Kengie >Few people, or no one, will take the time to see all the debug. >The key here is that the RTP port and IP negotiated in the SDP message >sent by asterisk to each party, should be "visible" for the party. A >common error is Asterisk sending in SDP a private IP address to a...

...pped RCF' in the GnuGK logs. Many thanks. Diagram of Network: External GK(Public IP)->GnuGK(Public IP)->Asterisk(Public IP)->SIP Phone(NATed) Also I would like to ask if it is possible for Asterisk to be listening on 5060 and another non-standard at the same time? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060208/d93f85c1/attachment.htm

...ppen when ooh323 module isn't registered to a gatekeeper. This current version i am using for the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version for the ooh323 and also how can i get the newer version of the ooh323(0.8.1) to compile with? Many thanks to you all. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060315/2244dcfe/attachment.htm

...th a sipfriend of type friend username=123 Call make from SIP Phone (username=123) to gateway, which sends to asterisk. The asterisk would use the context testing instead of from outgoing-call, since I assume that it is using the From field in the SIP Message. Please help. Many thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110329/a9a028a3/attachment.htm>

....193.155.2:15000 (type 3, seq 37417, ts 880320, len 33) Got RTP packet from 66.193.155.2:15000 (type 3, seq 5894, ts 899165, len 33) Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37418, ts 880480, len 33) Got RTP packet from 66.193.155.2:15000 (type 3, seq 5895, ts 899325, len 33) Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060118/d98e263a/attachment.htm