similar to: How to bridge 2 calls together

I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 #<< ==== I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Hello all, Hoping someone can help me with an issue...I have i .call file which calls out on a SIP channel and connects to an extension which dials another SIP channel. (both via voip providers) - both to PSTN. Problem is, hanging up the POTS phone doesn't release the channel (either one - hanging up the calling channel or the destination doesn't do it). Using IAX instead of SIP works

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

I've seen other posts about this problem, but I haven't found a solution. I'm dumping eight call files into the "outgoing" directory at one time. Three of the calls are successful while the other five are lost. Here is the call file: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Note: All

I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Using the "|caller" parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable

I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I have just about everything working except for the message waiting indicator. I have the following setup in context [ccm] in my extensions.conf file: ;MWI exten => _2807XXX,1,SetCallerID(${EXTEN:3}) exten => _2807XXX,2,Dial(SIP/28888@65.202.115.240) exten => _2807XXX,3,Answer exten => _2807XXX,4,Wait,1

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read

Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file

Dear Sir, I'm trying to configure the SMS capabilities of my provider (Telecom Italia) with my asterisk. I tryed with a normal phone (SMSC number 42100) and all is working fine, incoming and outgoung SMS. Now I'm trying to configure my asterisk for support SMS, the incoming messages are working fine (incoming number 042111) dialplan match and SMS recived. I'm getting troubles in

Hello, I tried to send sms for local extensions and i observed that file is created but sms isn't delivered yet. Can someone help me with this thing? rr:/var/spool/asterisk/sms/mttx # cat ../../outgoing/smsq.mttx.0.1322430026-20217.1 Channel: Local/1010 Callerid: SMS <1010> Application: SMS Data: 0,s MaxRetries: 0 RetryTime: 30 WaitTime: 10 rr:/var/spool/asterisk/sms/mttx # ls -la

Like some other people on here, I am trying to integrate Asterisk for VM with CCM version 3.x. I've got gnugk and Asterisk running, I've got CCM registering with the GK, I've got the voicemail pilot and profiles setup. A call comes into a CCM phone, it rings, rolls to the correct VM on ASterisk and asterisk emails the voicemail and I can check the voicemail, but I cannot get MWI

Hi all, I've been trying to get SMS operational on my Asterisk box, which has a TDM400P card with a pair of FXO interfaces configured (ZAP/1 & ZAP/2). I've not had luck with either of my lines, after issuing the command "smsq --motx-channel=ZAP/1/1709400X 00000 register". I see the following output in my Asterisk console: -- Attempting call on ZAP/1/17094009 for

Hi Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

Hi, all, I met a problem for several days, any suggestion is really appreciated!!! I'd like to do autodial using Asterisk. For example, I have a file under /var/spool/asterisk/outgoing, which include: channel: zap/g1/12345 MaxRetries: 0 RetryTime: 60 WaitTime: 20 Context: default Extension: 2222 Priority: 1 And in my "extensions.conf" file, I have [default] exten =>

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid