similar to: Multiple IP's (aliases) on asterisk box?

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

I'm about to download Festival source, apply 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

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 >

Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't

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

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

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

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.

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?

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Has anyone had problems with using hold on a 7960 SIP firmware? The problem is when the 7960 puts a call on hold and you take it off hold again, the 7960 outbound audio is delayed on the other end. Sometimes up to a few seconds. I've tried a couple different things, making the "other end" a diff type of trunk ie: 7960sip --> asterisk --> IAX2 --> PRI 7960sip -->

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

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".)

What's the proper syntax for implementing directed call pickup? Running cvs-head from today (9/24/05 including Mark's fixes), and tried: exten => *99,1,Pickup(${EXTEN:3}) but that does not seem to work, and there isn't an example in the configs directory. 'show application pickup' suggests the above should work with our sip phones, but apparently I'm missing

I've not been able to register with iaxtel.com for the last couple of days. Is anyone else seeing this, or did I miss something?

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as "unavailable" (not the 102 busy as

Anyone know of a business class sip hard phone that includes a quality display capable of supporting "push" data (maybe Polycom?). Something like... VM: 3 msgs OurStock (1:43pm): 59.5 somewhere on the display that can be updated (pushed) from a server? Rich

Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and