similar to: Onhold Music

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Anyone can share me with Music Onhold Configuration sample? Thanks in advance for your help, Kang

I'm trying to compile the last * CVS version and I got this error: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-09/10/03-18:47:18\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

----- Original Message ----- From: hank To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 27, 2005 9:13 PM Subject: shoutcast mp3 music onhold with amp portal? hello I use amp the asterisk management portal I went to www.voip-info.org I followed the instructions that were given on http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf it isn't working

There were a number of messages about using a digium card as a data t1 (router) kind of thing. Is there a howto out there or a status that I can see whether this is feasible or not? Or is it feasible but very difficult? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I'm running the latest version of cvs (not stable), I'm not sure what the other end is running and if this has been fixed or not yet, however I was playing round with onhold earlier, the call went to onhold, and came back from it, then 2 seconds later was hung up unexpectedly, below is what was on console... -- Started music on hold, class 'default', on

Hello All, I am battling a problem I don't know how to fix... Here is the scenario: Fractionated T1 with 1-6 channels voice, 21-24 channels data. Comes into a box with Digium T100P, splits off data channes with HDLC, to devicec pvc0, This works fine, but I have a routing problem getting to my machines! Here is the current set up... sethdlc hdlc0 fr lmi ansi sethdlc hdlc0 create 44

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

Anyone try the new Cisco firmware for the 79xx sip phones? In my test it seems to work fine for a little and than soon the phone looses its time. At first the status shows clear, and then it appears to get confused about the ntp time source and the time goes away on it. No features, just bug fixes. Perhaps I have something wrong with my ntp settings, but the other firmwares work just fine. --

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

After many months of development, I'm pleased to announced Firefly - an IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to multiple networks, skinable interface, connection to firefly network, IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here -

Here is a section of my dialplan (macro) exten => s,200,Wait(1) exten => s,201,read(holdopt|screen-onhold|1) exten => s,202,GotoIf($[${holdopt} = 1 ]?4) exten => s,203,GoTo(200) it's simple really it loops telling you the caller is on hold until you press 1 and then it sends you off to another area. The problem right now is that if the read() times out i get these warnings...

I'm trying to use fine axis controls in lattice for each panel. Specifically, within each panel, I want to set the limits for x and y equal to each other since it is paired data (using the max value of the two). Of course, I have no problems setting the limits for the entire plot but I am having trouble setting them for each specific panel. Could someone please provide me some guidance?

I'm wondering if I am dealing with a limitation in lattice. It's probably due to my own limitations though. I'm working with a lattice dotplot. The x-axis is set at "free". In one panel, there are only two data points that have equal x-axis values. Since no range exists, lattice isn't plotting those points. It reports: "Error Packet using 6 'x' and

Does anyone know if the 3rd patch listed on this bug fixes the seg fault problem related to voicemail? http://bugs.digium.com/view.php?id=4800 -- respectfully, Joseph --------------------

I have a User model and a Status model. Each user is given a certain status, either "Accepted", "Rejected", "OnHold", "Declined", etc. I need to find all users with a status of "Accepted", so I''m doing the following: in User.rb: belongs_to :status def accepted_users status_id =

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

I have just setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a standard 1.0.3

Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no success. i have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and =/usr/sbin/ast-playlinein