Re[2]: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call

2004-03-12 Thread Scott James Williamson
Let me start by saying I have no cisco phones, and no idea how to configure them. I will speak to the use of asterisk behind a NAT'ing firewall, which I believe to be your setup. Asterisk is pretty picky about how SIP and RTP packets are handled by a NAT firewall. Basically you need to maintain th

Re: [Asterisk-Users] voicemail not working with mysql !!!

2004-03-04 Thread Scott James Williamson
Hello atif, send an e-mail to [EMAIL PROTECTED] I know nothing about voicemail and mysql configuration -- Best regards, Scottmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.

[Asterisk-Users] NAT, Asterisk and SIP service provider (sipgate.de)

2004-03-03 Thread Scott James Williamson
Hello Oliver, okay, this was not easy and will make a long e-mail that I will also CC to the list. I will answer in English because it is my native language. I lived in Germany for 2.5 years and can speak German okay, however I will spare you all of the declination failures that I make on a regula

Re: [Asterisk-Users] SIP Behind NAT (sipgate.de)

2004-02-19 Thread Scott James Williamson
Hello List, Just thought I would post an update, I got asterisk to register with sipgate.de. I was wrong, it was my firewall (maybe). Here is the way a normal nat under openbsd pf works: udp 192.168.1.100:5060 <- 24.102.192.227:(random port) <- 217.10.79.9:5060 but add this line to pf.conf

[Asterisk-Users] SIP Behind NAT (sipgate.de)

2004-02-19 Thread Scott James Williamson
Hello Users, I am attempting to create a sip connection in the following network: Sipgate.de --> Internet --> Gate --> Asterisk PBX --> Some Extension Gate, the gateway and nat'ing firewall has sip udp (5060) traffic and rtm udp (8000 to 8020) traffic forwarded to the asterisk pbx machine. Both