similar to: Festival Installation - Asterisk 1.0-RC2 && Debian Woody

Hey All, I now have Festival compiled, installed and running using the instructions on the Wiki page. When I try to change the voice that is being used however, I am running into a problem. I get the following in the festival server log: Cannot open file /tmp/est_10877_00000/utt.wav as tokenstream Wave load: can't open file "/tmp/est_10877_00000/utt.wav" Cannot load wavefile:

I have loaded festival-1.4.3 patched with the 1.4.3.diff file. Festival source is in /usr/src/festival dir. When I try to use it I get this from asterisk: -- Executing Answer("SIP/chad-57a4", "") in new stack -- Executing Festival("SIP/chad-57a4", ""I am talking"") in new stack == Parsing '/etc/asterisk/festival.conf': Found

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

Following the installation directions on the wiki, I got festival built and installed. However, when I hit it from my dialplan, I get: Feature Token_Method not defined I found only one reference to this error message in the archives and there was no solution... Thanks! -Michael

Hello all, I'm looking for some help to try to understand why my CPE doesn't work good with Asterisk in MGCP. Here is what I want to do : - Register a TECOM AH4021 on Asterisk in MGCP with the following profile in mgcp.Conf : [general] port = 2727 bindaddr = 10.95.20.1 disallow=all allow=g729 allow=alaw 020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

I currently have an AGI that calls the Festival text2wave app to write a wav file that my dialplan plays into a call with the Background() command. But the voice sounds terrible: like SAM, the 1980s 6502 voice synthesizer. I tried to slow it down by calling (text2wav -eval "(Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 [...]), but it still sounds like it's talking while

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI> mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

I've been trying to figure this one out for a while, but to no avail. I have my cisco ubr924 setup for MGCP with Asterisk as the call-agent. I have manually registered the endpoint in mgcp.conf. When I pick up the phone, I get no dialtone and debug shows errors. IOS on the ubr924 is 12.2. Any help is appreciated. from mgcp.conf: [ubr924] host=65.37.86.203 context = from-sip (just as a

I know this is not the most appropriated list to this, but I will try: Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method? I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP,

Hi. I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk with the next configuration files. '--------------- extensions.conf ---------------------------------------------------- [general] static=yes writeprotect=yes [globals] ap1 => mgcp/aaln/ap200@64.76.148.186 [macro-apl1] exten => s,1,Dial(${ARG1},30,Ttmr) ;exten => s,2,Voicemail(u${MACRO_EXTEN})

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306

I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

New to asterisk so please be gentle. I'm guessing I'm among a number of recent additions to the list after the article in Linux Mag. I gotta say I'm *very* intersted in the project and will be doing lots of reading shortly. A couble quick questions first... How suitable is Asterisk for use as an IVR providing callers with textual data out of a database? Can it be combined easily