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Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Chameleon.1093614365.adar0@vegas> > Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 >...

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

asterisk trunk from April 1 on fc3. Box has been up for several months with no issues. Overnight, this remote box died, and rebooting shows the following on the console: exec of init (/sbin/init) Failed !!!: 20 umount /initrd/dev Failed: 2 kernel panic - not syncing: attempted to kill init Does this sound like a hard drive failure? The box is about 150 miles away and is inaccessible remotely.

Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near

I have a done google seaches on convertion and so far they all failed. Rich adamson and wheely-bin.co.uk Here is what I have Laptop running solarwinds tftp with the following files OS79XX.txt <- POS30201 SIP<phonemac>.cnf.xml RINGLIST.DAT <-Ringer1.pcm Ringer1.pcm The phone (7940) is hardcoded to the TFTP ip of the laptop and has a hardcoded ip is was on firmw...

Running fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rvvvvv). Would like to rotate the files without killing the cli session. Any reasonable way to

Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio

Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to "something" that the _majority_ can agree to call v1.1 Stable

...your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. RE: rtp channels not through asterisk (Brian West) 2. Turning "*" Hangup off in queues (usman@user.iphonica.net) 3. Re: Voicemail email notification (Rich Adamson) 4. Can't Make Outgoing Call (Norman Zhang) 5. Re: Voicemail email notification (Dorn Hetzel) 6. Re: Asterisk in parallel with PSTN [OT] (Rich Adamson) 7. Re: rtp channels not through asterisk (Rich Adamson) 8. Re: Realtime sipbuddies table structure why????? (Greg - Cire...

Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for "ivr menu".)

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No

...ply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

No, DON'T use mpg321, it will not work. Make sure you're using mpg123. See the wiki. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Rich Adamson > Sent: Friday, October 01, 2004 12:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MOH - 3 processes of mpg321 > taking 20%CPU each -normal ? > > > > I have P4 2.8 HT machine with Asterisk. I spot 3 processes > of m...

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases,

Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a "second" email address

Just installed fc5, installed correct kernel source, and trying to compile zaptel-1.2. Changed the link in /lib/modules/2.6.15-1.2054_FC5 to point to /usr/src/redhat/SOURCES. Like: lrwxrwxrwx 1 root root 23 May 12 15:21 build -> /usr/src/redhat/SOURCES A 'make install' still complains with: make -C /lib/modules/2.6.15-1.2054_FC5/build SUBDIRS=/usr/src/zaptel-1.2 modules make[1]:

Guys. I would like to hear tips and tricks on extention config best practices, for example, naming, etc. and most of all, how to deal with extention that have a full time hardphone configured and assigned and then a softphone connecting to the same extention, for example, one employee has its hardphone on the office but sometimes when he travel, he uses his softphone to work with, what happens

For those that have experienced low VM recording volumes when using a Digium TDM04b (or similar analog pstn card), a work around has been committed to cvs-head. Need some folks to test it; it doesn't seem to work for me, but need some feedback from others to ensure the work around is actually functioning. (The work around relates back to bug #2023, and was committed around Thursday or Friday