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Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

...erface and if you ever > press the eject on the IPK you could use the t-1 as a PSTN interface. > > > --__--__-- > > Message: 3 > Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > Date: Mon, 19 Jul 2004 16:28:25 -0700 > From: "Wiley E. Siler" <wsiler@e2020inc.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@lists.digium.com > > Mine does the same. Once in Message center I can choose selection > 1.Message Center and then soft key Select. Then I select the > registered line that I want to check voice...

I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk don@moskaluk.com www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date:

Hi All Just received my first 7905G from a distributer here in Sweden. According to the spec this phone should be able to use SIP. Now I been looking on Ciscos home pages for several hours trying to find a "SIP image" for this phone. No luck at all, need special access to be able to download software to this phone. Is it the fact, that I have to pay for a contract of some kind to be

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

Is there a user guide for Comedian Mail? I need to give some training materials to my end users. So far, I have been unable to find anything through google or the Digium site. THanks, Wiley The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other

I ahve been searching to no avail for a referenc eon how to setup a part of my dial plan that will ring certain groups of number based upon the context. Essentually, I want to be able to designate 3 people as sales and have my IVR handoff and ring their extensions in order. Then maybe I will ahve a couple of people I group together and have them ring if someone selects 2 on the IVR for tech

I attempt to run make clean:make install and I get the following (cut short for brevity). zaptel.c: In function `zt_init': zaptel.c:6123: warning: implicit declaration of function `register_chrdev' zaptel.c:6124: `KERN_ERR' undeclared (first use in this function) zaptel.c:6124: parse error before string constant zaptel.c:6129: `KERN_INFO' undeclared (first use in this function)

When receiving an incoming call, I get sent to my IVR just fine. My Playback event plays back my test file and the itis suppose to Hangup. The Hangup app fires off and I see the console say it hungup the line. However, I cannot receive anymore calls after that. When I run 'zap show channel 1' I see my zap channel in a state of off hook. The only thing that fixes it is stop/restart

MP3s have to use constant bitrate not variable bit rate. Look in the documentation for mpg123. -----Original Message----- From: Jozeph Brasil [mailto:jozeph@globalmedia.com.br] Sent: Saturday, July 24, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: RES: [Asterisk-Users] Play CD! I do that. But when I play MusicOnHold the music is played slowly! I don?t know why... but is how

A better explanation can be found here... http://www.digium.com/index.php?menu=faq#TDM%20&%20Analog_0 > -----Original Message----- > From: Benjamin on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: Monday, September 27, 2004 11:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FXO question

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

You should see something like this.... (I have 8 channels) tuxpbx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo incoming en 1 incoming en 2 incoming en 3 incoming en 4 incoming en 5 incoming en 6

Just to make sure this isn't a typo in your original email... Is this example from your zapata.conf? Also, the extension you have shown are in extensions.conf not zapata.conf correct? Here is an example of a good zapata.conf.... [channels] language=en busydetect=yes faxdetect=both busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes