similar to: CallerID Advice ??

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", "") in new stack -- Executing [6004 at

The problem I was experiencing is still occurring, and it is getting worse. There are several names that Festival "gets stuck on". I don't know if it is a Festival problem or an Asterisk problem. The scenario, a call comes in goes through the dialplan (shown below in original message), and either reads the value ${FULLNAME}, but then hangs. It doesn't go to the next command

I?ve got an interesting problem and am wondering if anyone can shed light ? I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya Definity G3R via a Digium TE220. Asterisk 1.4.20 Zaptel 1.4.4 Libpri 1.4.4 MySQL 5.0.45 Festival Speech Synthesis System: 1.95 We have about 4200 accounts in a MySQL db. Asterisk retrieves the user information from the database, festival tts says

Hello-- In the interests of playing around and wasting time, I've installed the latest version of the Festival stuff, 1.95beta. And, in the interests of future Asterisk-Festival connectivity, I applied the 1.4.3 patch to put in the asterisk related routines. I did it by hand, but, it looks like the patch will apply with no comment. Asterisk works with the new server... BUT the speed of

I'm having trouble setting callerid with teliax. I use a simple dial-out subroutine to set the callerid depending on the calling extension, and then dial out. Teliax is saying they're not seeing any callerid info. [DialOut] ; subroutine for dialing out. exten => s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} ) exten => s,n,NoOp(${CALLERID(num)}XXXX) exten =>

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

This is probably a good time to point out that there is a good litmus test for all Voip providers. PRIOR to purchasing anything, send them an email and request the sales information. Ask about their servers or their policies or anything you can think of. How they respond will tell you a lot. If it takes forever, you can tell that they are either really busy, really indifferent, or something in

I've tried to create a subroutine that sets callerid name based on number. extensions.conf: ........... exten => s,1,Answer() exten => s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) exten => s,n,Dial(${mainline},60) ....... [set-callerid-name] exten => 0,1,NoOp( no CALLERID num set) exten => 02025462677,1,Set(CALLERID(name) = "Fred" ) ................ exten =>

Hello. I'm having trouble with callerid on outgoing calls. The recipient of the call only sees "unknown" rather than the number I'm specifying. If I set callerid info when calling an internal extension then I see the callerid name and number when I call that extension. I did that thusly: exten => 101,1,Set(CALLERID(number)=1112223333) exten =>

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding

I'm looking for suggestions for a good IP phone to use with Asterisk which can display CallerID information on the phone's LCD panel. I am currently using a Sipura 2000 for all my analog phones and I have purchased 1 bugdgetone 100 IP phone. I find the voice quality of the Sipura 2000 to be better than that of the budgetone. I find myself wanting to use the analog phones instead of the

Is anyone successfully passing Outbound CallerID to Teliax? If so can you please tell me how. Thanks in advance! Dan

hi; i have a customer who uses freepbx. He likes the ability to request a caller's name and then pass that threw to the extension, call screening, however one thing he wants to get is the ability to just pass the callerid to the extension, verbally red out to him. Can this be done? He's using latest build of freepbx, and i am as well. I'll do testing on it before deploying to

Just and FYI. In RPM form, works on 4.x and I will try on 5.x too. Speech is tacky, but if I can get something but the Gnome Festival synthesizer, maybe it will improve? http://www.adobe.com/products/acrobat/readstep2.html?promoid=BONRM -- Bill

If you asked this question on the biz list you would get a lot of people that will tell you that they offer services where you can set the caller ID to what ever you want. To name a few:: Nufone Teliax Voipjet ----- Original Message ----- From: "Doug Crompton" <doug@crompton.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Greetings! I was wondering if it is possible (using something like a group of Sipura SPA-841 IP phones) to have * announce information about the calling party via the SPA-841's speaker to a selected set of extensions that aren't set to "Do Not Disturb"... i.e., have * say the number, or perhaps have Festival speak the name, etc. If so, any hints and tips on how you'd go

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated.... sooo close.... TIA! -Ron [3004] type=friend username=3004 password=XXX

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port