similar to: Inbound IAX2 calls has no music on hold

Pardon the newbie post. I installed Asterisk on a "test" system using the Asterisk@home cd image. When you boot from Asterisk@home is installs an O/S and Asterisk on your PC. How cool is that. But I was wondering if someone could point me in the right direction for updating the version that I have. I'm new to CVS, how do I determine what version to build? Is there a primer on how to

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve szmidt Sent: Monday, June 27, 2005

I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt "They

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, A thought occurred to me; Background; In the early days of cable, the cable people seemed clueless to things like over selling bandwidth. But as time went along they got it better and better under control. Of course their natural competitor, DSL, created a bigger demand for them to get things under control. Today cable is often giving

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up

I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? I ask, because I spent yesterday giving it a whirl, and everything seems to go just fine till the very last minute. Zaptel, libpri and asterisk compile just

Hi there, I just tried today's CVS: 4/23/2004 version and found a strange loise with music on hold. Basically, when on hold you hear very distorted music as if it was very loud. This is the exact same problem described last year at: http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html No answers on

Great points Steve. I think the best we can do is all throw the newbies a bone ounce in a while. Redirection to the content that is relevant is enough to get most people on the path. Like you said, the hardest part is not seeing the trees for the forest. This is the whole "teach a man to fish" parable. It is pretty easy to tell someone A) How to search and where to look B) The

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/9e69a5c9/attachment.htm>

I'm running Connected to Asterisk 1.4.18-1. We have a T1 PRI and a SIP trunk coming in from our IVR. When people call in on the SIP trunk, I experience the following problem: I put the caller on hold, MOH starts. MOH quits after approx ten seconds. When I pick the call back up off of hold, they can't hear me for about 20 seconds (sometimes longer or not at all) but I can hear them. This

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen this

Hi there, We're experiencing the following problem on a handful of boxes and I'm wondering if anyone has any pointers for me. First up, hardware configs on boxes are the same: Dell 860, Dual Core Xeon, 1Gb RAM, 2x 160Gb SATA's in Software RAID 1. Various Trixbox versions - 2.2.3 and 2.4.2 to name two of them. Problem is that occasionally the music on hold speeds right up -

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten