similar to: Asterisk left in a bad state

Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half expected Monitor to use something like ZAP-2-1-<yyyymmddhhmmss>-in/out.wav for it's

Mahmut, excellent summary :-). I look forward to your next update. One little thing, In the manager events that show start/stop monitoring, can you please include a field that indicates the filename(s) to which the monitoring was written? Thanks, Ben -----Original Message----- From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] Sent: Tuesday, April 15, 2003 5:17 AM To:

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

Thanks Ben, Adam and Petr for the feedback! So currently things that need to be done for the Monitor resource are: 1. Name files uniquely. Adam, your naming suggestion is great. I think we should stick with that, with a minor change: I don't think we should put destination channel name in the file names. In some instances there will be no destination channels (plain IVR: play, record, dtmf),

Mahmut, First of all, I'd like to reiterate what a great patch this has been. I'd also like to voice support for having an option to mix the files on the fly, and name them uniquely. While I was able to smoothly put the files together with soxmix, I see on the fly mixing as hugely beneficial to an automated solution to saving/delivering the messages without intervention. One feature,

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi, all Please help me. My platform is RedHat Linux 9.0. I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk typing # asterisk –vvvvc I get one error and one warning: [chan_iax.so] => (Inter Asterisk

Our company manufactures an E1 channel bank that is approved for use in Australia (it should also be compatible with Euro standards). It is modular and available in 10, 20 or 30 analog port configurations. Signal monitoring and configuration is via Ethernet. These units are manufactured in low quantities for specific telco requirements. However if there was enough interest, we would be able to

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hello-- Why do I target chan_sip for so much effort? Because, it seems to me, chan_sip is probably the most used channel driver in the asterisk community!! (and, of course, the zap/dahdi driver, is also pretty popular) I haven't had time to follow up on chan_sip, and I probably won't for several months. But, if I had time, here is what I'd do: There are two ways to speed up

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: > All switches and routers give

G'day list, I am getting a lot[1] of traffic on my Internet link, ICMP messages from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address belongs to iaxtel.org). I see from the wiki that IAXtel supports only IAX2 from December 2003. Fine, however it looks like my * still wants to try and register using IAX1, and I can't find how to turn this off. This situation is