similar to: transfer from sip to dahdi, connects caller to MOH stream and not target

I have had this problem for a while, so I can't be sure when it started or what was changed. The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP,

Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? -- Regards, Shariq Khan 0333-3501125 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/ca6cfc50/attachment.htm>

We just finished an upgrade of our Asterisk system to an HA environment on a pair of servers using Linux-HA. As part of the upgrade, we also moved to Asterisk version 1.8.4.3 Most things are working quite nicely on the new system. However, I?m having trouble getting a paging feature to work. In Asterisk 1.4, we simply used the Page() application like this:

I get MOH when the caller hangs up. Is there any way I can just get Busy tone. Regards Michael Knill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060522/4905a84b/attachment.htm

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provider Marek

Can anyone shed some light on what happened? Asterisk 1.2.1 with Zaptel 1.2.1 Here is what I know happened: A call came into our main number and was answered Asterisk set the monitor CALLFILENAME and then started monitor. The call was directed to a context called "open" where all calls go during business hours. The dial plan has a Answer() again, and then played a message (custom/1)

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

Most of the time i can find answers to my questions on the wiki, google, or searching the list now i am stuck . I have a small * box at my house running 1.0.9 stable and a devlite kit. Every thing is awesome VM, IVR, Echo canceling, and Meetme are all working great. But on Incoming caller id i need to add a 9 as a prefix to make it easier to return call from my cordless phone (cheap vtech

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

Hello, i?ve a problem i dont find the reason for. An incoming call coming over iax is connected to a Sip phone. Until the phone picks up the call i could hear moh without problems. Then the phone sets the call on hold and opens another call to another extension. The incoming call hears the Hold music and also the call to the other extension hears another moh. Everything works so far as it

Hi there, Is it possible to start a new MOH stream for each queue member? My queue MOH is a message rather than music and I want customers to enter the queue and hear it from the start rather than from some random point. The first person entering the queue always seems to hear it from the start. I know could (ab)use the position announcement settings (which interrupts the MOH), changing the

Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered

Hi, my Problem ist he following situation, Caller A calls to my company. HE gets into my call queue and is then answered by caller B. Caller B answers and makes an attended transfer to caller C. Caller B hangs up bevor caller B has anwert the call. Caller A can hear MOH till caller B hangs up and gives the call to caller C. Then Caller A could only hear the free ringing sign.

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

Hi, I have some customers complaining about moh quality, but never about voice quality. The moh files themselves sound great when listen to from a media player, and given that I am streaming the files through DAHDI (while voice goes from DAHDI to SIP, so more chances of problems) I would expect moh to be of more consistent quality than voice itself, but it turns out to be the opposite.

Hi, Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan. This is what I have in chan_dahdi.conf: [channels] language=en echocancel=yes usecallerid=yes cidsignalling=v23 sendcalleridafter = 2 hanguponpolarityswitch=yes rxgain=2.0 txgain=3.0 progzone=uk

2012/1/10, Olivier <oza_4h07 at yahoo.fr>: > Hi, > > 1. This patch didn't correct the issue but I'm far from certain that I > correctly applied the patch. I was right to suspect I was wrong : now, after correctly applying the DAHLIN-275 patch, it's working OK (with the EchoCan module plugged-in). Thanks for your lighting fast correction !! > 2. I took the

Hi, We have a system with both ISDN trunks and SIP. We receive incoming calls on both but always dial out via SIP. When dialing out the caller id is set like this: exten => _X.,1,Set(CALLERID(num)=${CC_ORIGNUM}) exten => _X.,n,Set(CALLERID(name)=${CC_ORIGNAME}) exten => _X.,n,Dial(${CC_DIALSTRING}, 60, em) This always works fine on SIP and on ISDN as well when the number is not hidden.

I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). -- Executing [7000 at chambers:1] Park("SIP/2000-00000008", "") in new stack == Parked SIP/2000-00000008 on 7001 (lot default). Will timeout back to extension