Re: [Asterisk-Users] Dialing 800 numbers with VOIP

On Tue, 10 Feb 2004, Chris Clifton wrote: I'll second this. For the past 4 days, Vonage can't figure out how to process our visa check card. In the meantime, Nufone has us setup with an account, ready to roll. - Chris Clifton Interesting. I've been trying to get Jeremy to set up a second

[Asterisk-Users] Spanish indications configurationº

Hi all We've been using * for a while here in Spain, but some people has told us that they have problems when they type an extension calling to us. I've been trying to find out what's going on, and it's an issue that only happens with some ISDN and analog calls, not from mobile calls as

Re: [Asterisk-Users] x101p beeps/sceeching

On Thursday 12 February 2004 18:11, Jeff Gustafson wrote: I'm experiencing periodic beeps or screeching when I'm on a call via the x101p card to/from PSTN. Echo cancellation seems to be working fine. The beeps seem to happen with echo cancellation on or off. Is there a setting I can

[Asterisk-Users] Digium connectivity issue?

Are others seeing hugh delays and/or lack of connectivity to Digium? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] GS BT-100 echo

I picked up a GS 100 phone based on the overall good response I've heard of these phones. One thing I'm fighting with, which I can't find any info on, is a *real* bad local echo on the GS. The remote end doesn't hear it, and all the docs I see about echocancel deal with hardwired phones/ports

[Asterisk-Users] chan_local and variables

We need to implement the following: Call comes in, ring ZAP/1 (6 rings) For the last two rings, also ring ZAP/2 I have the following (which works as expected): [incoming] exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18) [test1] exten = 123,1,Dial(ZAP/1)

Re: [Asterisk-Users] Solved! x101p beeps/sceeching

It turned out that a Gb/E card was too close to the modem causing activity on the card to bleed over to the modem. ...Jeff On Thu, 2004-02-12 at 16:11, Jeff Gustafson wrote: I'm experiencing periodic beeps or screeching when I'm on a call via the

RE: [Asterisk-Users] Direct mailbox transfer

Thanks guys I will try that in the morning... Sean -Original Message- From: John Fraizer [mailto:[EMAIL PROTECTED] Sent: Thursday, February 12, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Direct mailbox transfer Sean Garland wrote: How would one implement a

Re: [Asterisk-Users] More external call control

On Thu, 12 Feb 2004, [EMAIL PROTECTED] waxed: My questions are as follows, (but before I begin; I know there is queueing and some ACD functionality in *, but I need to do this externally. I want the queueing decisions to be external because my central queue engine handles things like email,

[Asterisk-Users] 800 numbers / Skinny - IAX2

Hi Folks, I'm trying to route IAX2 calls to 800 numbers from a Skinny channel and the log says: -- IAX2[69.73.19.178:4569]/4 stopped sounds -- Asked to indicate 'Stop tone' condition on channel Skinny/[EMAIL PROTECTED] And no audio happens. It's working for Zap and SIP channels, though. Tried

[Asterisk-Users] Hide outgoing CallerId on Zap interface

Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would

Re: [Asterisk-Users] festival voices

(Parameter.set 'Audio_Method 'linux16audio) ;(Parameter.set 'Audio_Method 'esdaudio) ;(Parameter.set 'Audio_Method 'mplayeraudio) ;(Parameter.set 'Audio_Method 'sunaudio) ; American female I'm using the cepstral frank with festival ;) (set! voice_default 'voice_frank) in /root/.festivalrc bkw

Re: [Asterisk-Users] Anybody going to the Spring VON converence [OT]

I'm going. Would be great to have an Asterisk gathering. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] channel bank - Adit 600

hi! I would like to check the apllicability of Adit 600 and Adtran 750 in converting FXO + FXS to E1/T1 channel to be sent to a * voip box. we r currently using pleidaes channel bank and it has the problem FXO lines hanging forever.(Don't disconnect). Bundle of FXOs are obtained from inhouse

[Asterisk-Users] Voicemail Password Digit Timeout

I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail

[Asterisk-Users] Codecs compile error on yellowdog

greetings I'm running yellow dog 3.1 compiling Asterisk 0.7.1 during the make process it seems to die at the GSM build. (summerized) As build goes' through must remake `src/add.o'. entering dirctory `/usr/local/asterisk-0.7.1/codecs/gsm' gcc -march= -fomit-frame-pointer -c

[Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones

Hi, Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. Any ideas would be helpfull. Thanks

Re: [Asterisk-Users] System freeze

On Thursday 12 February 2004 16:34, you wrote: On Thu, 2004-02-12 at 12:28, B. J. Bomar wrote: I too have seen a couple of system freezes for no apparent reason. I am * on a RH9 box with kernel 2.4.20-28.9. Without wanting to sound like a RH basher that I normally am, could this be a RH

[Asterisk-Users] Re: Codecs compile error on yellowdog

I must be doing something wrong i have installed gsm.rpm manually and tried to recompile, but i still get the same error. make[1]: Entering directory `/usr/local/asterisk-0.7.1/codecs' make -C gsm lib/libgsm.a make[2]: Entering directory `/usr/local/asterisk-0.7.1/codecs/gsm' gcc -O6 -march=ppc

[Asterisk-Users] HELP!!!! Having problems Starting Asterisk

I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I

Re: [Asterisk-Users] Direct mailbox transfer

My extensions start with 7XXX [ext-direct-to-vm] exten = _67XXX,1,Wait(1) exten = _67XXX,2,Playback(/var/lib/asterisk/sounds/voicemail/YOUR_CONTEXT_HERE/${EXT EN:1}/greet) exten = _67XXX,3,Voicemail2(${EXTEN:[EMAIL PROTECTED]) exten = _67XXX,4,Playback(vm-goodbye) exten = _67XXX,5,Hangup

[Asterisk-Users] Adtran 750 - what do I need

Could anyone tell me what I need to include in the purchase of an Adtran 750 to work with a T100P? Obviously, I'd need a combination or FXO and FXS boards to fit my application, but, are there any other boards that are required? Does every Adtran include the proper port to connect to the