similar to: Voice mail volume

Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten => 2211,1,Dial(SIP/211,10) exten => 2211,2,VoiceMail(u211@default) exten => 2211,3,Hangup Here is the relevant part of

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

Hi: Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server isn't seeing the mainboard's APIC. -Stephen-

On Sat, 2007-07-07 at 08:39 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Fri, 06 Jul 2007 12:02:53 -0600 > From: Stephen Bosch <posting at vodacomm.ca> > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID:

Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere

Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a "ring group" and "hunt strategy" How can it be done

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some

Hi, folks: I remain intrigued by the gap in BRI implementation between North America and Europe, and I wanted to get feedback from the list members on the matter. I'm seriously considering making the leap in our office. In Europe, the idea that an office that does not have enough lines to justify PRI would use analog lines is perceived as technologically backwards, and yet that's what

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-

Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

An outside caller started to leave voice mail. The CLI shows: Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: gsm, 0x8295d40 -- x=1, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: wav, 0x829e2c0 -- User cancelled by pressing 0 -- Playing 'vm-saveoper' (language 'en') Later on,

Hi: What's the trick for getting the Polycom IP 501 message light to go on when there is voicemail waiting? Any ideas? -Stephen-

Hi All, I would appreciate a lot if you could help me. I have installed Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have also installed 1 FXO port card: Digium TDM400P. After loading zaptel driver I could see my digium card's led glow green. Tested with zttool that its in OK state. I have configured fxsks=4 in zaptel.conf(channel 4 cause FXO module is on port 4). I

Hi All, I have an issue on voice mail with odbc in asterisk 11.7 box. Voice message can be received through Google mail but it doesn't show in phone. The error messages is as follow and let me get your kind advice. -- <SIP/0015-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28 14:55:13] DEBUG[12260][C-00000006]: app_voicemail.c:3824 last_message_index:

I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

Hi: I have a Sangoma A200 card installed in a server with two FXO modules and one FXS module. Analog sets connected to the FXS module have a "squeaky static" -- it's like static mixed with the sound of someone vigorously cleaning a window a few doors down. In other words, it's not a classic static noise, but it is noise, and it's distracting. Remote callers can hear this

Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks,