similar to: Still can't pickup parked call

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 =>

I have a simple asterisk install (1.4.18), and want to use call parking. I can successfully park a call (I see on the CLI that the call is parked to 701). Everything is pretty default. However, I can't pickup a call from another phone. When I dial 701 from a phone, asterisk can't find that extensions and notifies the person picking up that the extension doesn't exist. (It appears

Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in).

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.

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

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]).

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

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 using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

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 list, sorry if this has been discussed in the past, but I couldn't find anything wise about it. Since we had some trouble with the builtin hold function of some (all?) SNOM 320/360 phones, we decided to use the call parking feature in asterisk instead. Assume, a call comes in with CALLERID(num) 1234567 for extension 10. Extension 10 parks this call into 801, dials extension 11 and

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

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

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.

svn trunk 31497 For the life of me, I can't get this :) I want to be able to catch the situation where the calling party hangs up *before* the call is connected to the called party. My dialplan is thus: macro DialExternal(exten) { Dial(Zap/G3/${exten},120,g,M(connected)); goto DialResult|r${HANGUPCAUSE}|1; Hangup(); }; But the goto dialresult is not executed: Executing

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to "yes" in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think this is probably the right track though. Any insight would be much appreciated.

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");