similar to: Events when the target answer

Hi Group, I am sending my question again why I don't have answer yet: I am developing a application, this use "Manager API" to connect with Asterisk. But when I call to an external number (over a zap channel), I don't receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel -------------- next

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time I want Asterisk to delay answering the POTS line via a

I want Asterisk to delay answering the POTS line via a Wildcard (a Zap channel) by some period of time, either a number of rings or just a number of seconds. I have tried this: [from-pots] exten => s,1,Wait(30) exten => s,n,Answer ... exten => s,n,Dial(SIP/brian&SIP/joe,10,H) exten => s,n,Voicemail(u2001) exten => s,n,Hangup exten => s,103,Voicemail(u2001) exten =>

Hi everyone, I''m Ezequiel from Buenos Aires. I''m new in this list, so please forgive me if my questions are not the correct one''s fro this alias. I''m working as a campus ambassador in Sun, while finishing my engineer degree, creating communities around opensolaris in Argentina. I discovered this project, and this group, looking for something that I could

Hello All I have a TDM2400 card with 4 FXO, and the the following problem: This card answered all the calls, but for the caller, the call is ringing and I don't hear nothing when it has picked up. Here is piece of my log: Thanks for any help. == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' -- Executing [s at entrada:1]

Location = US asterisk/zaptel from CVS. Updated last week some time. Currently rebuilding with todays checkout. I have 2 fxo channels hooked up to outside standard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel => 4,3 Then any call routed from asterisk to the outside line will ring, and can be picked up, but *

I have this in extensions.conf: exten => 1,1,Dial,Zap/g9/4439568899/|24 exten => 1,2,VoiceMail(u8004) and this happens: -- Playing 'js-joe-trvm' -- Executing Dial("Zap/2-1", "Zap/g9/4439568899/|24") in new stack -- Called g9/4439568899/ -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Nobody picked up in 24000 ms -- Hungup

I have installed on a VirtualBox vm the crossbow ISO Now I''m trying to create the network map described in the "getting started" pdf I have this situation: On global zone, I have the real NIC: e1000g0 with ip 10.0.2.15 (VirtualBox NAT this IP, I can get web access with it) I have created an etherstub (etherstub n?2) with 2 vnics (vnic5 and vnic6), I''ve plumbed

I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX

Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set:

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In

Hi, I'm having trouble with dialing out on analog lines. Asterisk can't seem to detect "answers". I have two zap groups. Group 1 is connected to an external analog PSTN provider. This group seems to work fine, especially with "answeronpolarityswitch". Group 2 is a group of "GSM gateways", ie. devices that host SIM cards so that you can dial from any PBX

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm using a TDM400 with one FXS and one FXO module (developer kit) and I've been testing termination from SIP phones to PSTN and it works fine, but asterisk accounting is doing something strange (for me). Scenario: 1 - extension 1009 (SIP phone - BT101) 2 - Zap/4-1 (TDM400 FXO module) extensions.conf: [dialout] exten =>

People, My goal is to get asterisk dialing out via my landline (POTS) from a sip softphone. Ive got the phone, The TDM400p is installed and working. (See below) When ever I dial a number that is directed to the outgoing port on my card (fxs/fxo?) I get no ringing, then it claims its been answered. the CLI reports the following: Executing Dial("SIP/301-f97a",

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

Hi there Background: I want to add DDI and voicemail to users on an existing analogue pabx.. It does not support ISDN. I have 10 DDI numbers via IAX which I am having sent to my Asterisk box. I have 2 X100P cards connected to 2 analogue extension ports of my main legacy analogue pabx. I have set up voicemail for each of my DDI numbers, and when a call comes in for the person at pabx

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

Hi people, the following bug makes some error messages to appear al log level 0 when they are supposed to be al level 3. Any guy at the printer team could tell if I am safe printing with all those messages and why they are generated? Only a few computers are affected in my network. https://bugzilla.samba.org/show_bug.cgi?id=10118 Thanks! On Oct 25, 2013 1:21 PM, "Ezequiel Larrarte"