similar to: Ast 13.11.2 : bridgepeer variable empty ?

Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: > Hi, > >

Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26]

Dear All, I'm very new Gluster FS user.... ...and apologize for my very poor english! :-) I have installed Gluster FS v. 1.3.12 on 2 server with Debian 4.05, in AFR with client side replication, explained in this page: http://www.gluster.org/docs/index.php/Setting_up_AFR_on_two_servers_with_client_side_replication Tested to work! :-) Now, my idea: Use a Xen Hypervisor with image disk

I'm wanting to capture the zap channel that a sip channel has connected to. I came across the ${BRIDGEPEER} variable documented on the wiki, and if I show channel SIP/<channel> when a call is connected I can see BRIDGEPEER as one of the channel variables. However ${BRIDGEPEER} is not set when I want it: I run a macro when the call has been connected. Does anyone have a hint on how

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's "Direct Bridge" variable. I have

Hi: I want to make parking calls easier for my hard-working users. Is there a way to make the Polycom call park feature work with Asterisk? In case it just works out of the box, I haven't tried it yet; but the "call park" feature isn't enabled on the Polycom phones by default. -Stephen-

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi, my question is how can i get channel-id of a user or peer. I tried using ChanIsAvail(username). this works correctly when user and asterisk are on Local LAN. But my asterisk server is on public ip and users are behind nat, and so this method says unknow host when used on public asterisk server. I also tried built-in variable ${CHANNEL}, but this returns the channel-id of the calling channel.

Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains "Tt". Then an outside call would operate as a Local channel in an internal context after the first transfer. If the

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hi, I have a question... With the variable ${CHANNEL} I can get the channel whose made the call, or the caller. How can I get the channel of the called side? Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

I''m looking to use an external node classifier (ENC) in our environment. What''s the easiest way to programmatically get currently applied classes (and class parameters) for all hosts, with the goal of dumping it into a database for later retrieval by the ENC script? Nodes are currently classified via site.pp. thanks -- You received this message because you are subscribed

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

Hi, we are experiencing significant performance degradation after live migration of hvm domains in Xen 4.0 (SLES11 SP1): after live migration the performance is dropping to less than 90%. I did a backport of cs 23420-23423 and the performance is okay now. I would like to request to include these changesets in 4.0 and 4.1. The backport is quite trivial, I can send patches if you are willing to