search for: lomonaco

Displaying 6 results from an estimated 6 matches for "lomonaco".

2006 Dec 22
2
System Application with java
...gt; /root/log.txt) exten => 666,4,wait(10) exten => 666,5,Playback(my-sd) exten => 666,6,Hangup And here the logging by Asterisk.. Connected to Asterisk 1.2.13 currently running on fedora (pid = 1951) Verbosity is at least 3 -- Remote UNIX connection -- Executing Answer("SIP/lomonaco-0945fd18", "") in new stack -- Executing System("SIP/lomonaco-0945fd18", "/root/example2.sh >> /root/log.txt ") in new stack -- Executing System("SIP/lomonaco-0945fd18", "echo "SUCCESS" >> /root/log.txt") in new st...
2003 Sep 26
3
RES: RTP routing..
...ynamicDnsClient and IPTABLES... I?d like to know if is possible to using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn?t receive sound... Andre Lomonaco -----Mensagem original----- De: Low, Adam [mailto:ALow@Prioritytelecom.com] Enviada em: Friday, September 26, 2003 9:06 AM Para: 'asterisk-users@lists.digium.com' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions:...
2003 Oct 15
4
indications.conf
...modprobe wcfxo Notice: Configuration file is /etc/zaptel.conf line 128: No such tone zone known: br 1 error(s) detected /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed Any tip to solve this problem... Thanks a Lot... Andre Lomonaco
2003 Sep 26
4
RTP routing..
Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that
2007 Dec 15
0
Open ITU G.107 Implementation to measure voice quality
Hi, Does anybody know where I can find any open source ITU G.107 implementation available? I'm looking a way to measure the voice quality in my projects.. Thanks in Advanced, My Best Regards, Andre Lomonaco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071215/7aec933e/attachment.htm
2007 Feb 11
2
TDM02B not working
I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400