similar to: when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am hoping for a little help from the collective. Inbound I have 2 different numbers. I can call in on both

So, I can't figure out how to get the parkandannounce application to work the way I want it to... I have cisco 7960 IP phones using SIP, and this is what I have in my extensions.conf: exten => 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1) exten => 700,2,Hangup and in my parking.conf: [general] parkext => 700 ; What ext.

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

I know the call parking feature changed in 1.4.23.1 to fix some serious issues. I'm seeing a major change though which I find disturbing. A person parks a call by transferring it to the parking position (700). When the timeout value is reached, the call is NOT returned to that device, but rather the 's' extension of the phone's registered context (in this case [unrestricted]).

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

Hey All, I am working on a SIP Call bridging application. Every time I receive a incoming call in Asteriskserver1 my AGI should alert to AsteriskServer2 and AsteriskServer2 should callback to AsteriskServer1 and call should be bridged on specified extension. (making call in this way is customer requirement) Every time I receive a call in AsteriskServer1, I Park the call and through AGI,

Hi, I am having a very weird problem with call parking. I have defined call parking correctly, as it work well when parking calls and picking them up. The problem is what happens after the the 45 seconds have expired. The behavior wanted is that the person who put the call on "park" is called back after 45 seconds. What ACTUALLY happens is that the phone who got put on park

I'm still struggling to pickup calls. I now have a single context (entryocginternal) where I have "include => parkedcalls". The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and

Using the default features.conf setup, if I include parkedcalls in my dialplan, and a call gets parked, and times out, where does the call go? Does it go to a timeout extension in parked calls, or does it go to a timeout extension in the original context? (Using an AEL based dialplan similar to below). -- context internal { ... ... t { jump 600 at

I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem.

Need help with how to configure for parked calls in the Flash Operator Panel's op_buttons.cfg file ... I've looked on the wiki, google and asternic's site and can't seem to find how to setup op_buttons.cfg to monitor parked calls. For example, if someone parks in 701, I'd like to see that represented on the panel. I've tried a number of things ... this is what I have now

On 25 May 2004 at 9:50, sdgesa gaeharth wrote: > I was using windows media player (with the patch for playing ogg). You mean you're using it for ogg-streaming? If I remember correctly just a few mails ago somebody said it only plays static files. Which plugin (patch???) are you using? Regards, Stefan --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project

100% repeatable (for me). Sip phone A calls Sip phone B. Either Sip phone A or B does #700. The party that keyed #700 gets the parked announcement (eg 701) and the other party get MOH. There is still an audio channel between the two SIP phones at this point. When the party that typed #700 hangs up, Asterisk crashes. This has been working in previous 1.4's (but not 1.4.10) and I