similar to: URgent - capturing 'answered'

Hello, I've been looking for this on voip-info and this list threads, and I am guessing I am not looking right. What I need is the way to capture (and write to DB) the information on who 'picked' or 'received' the incoming call. Here is the example of .rb file that is called from extensions.conf: private def lokal

Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if

On Dec 10, 2014, at 1:52 AM, Steffen Kaiser <skdovecot at smail.inf.fh-brs.de> wrote: > >> Global scripts were compiled: >> >> /usr/local/etc/dovecot/sieve # ls >> 10-move-spam.sieve 10-move-spam.svbin > >> However, I ran sievec again and tried saving a modified script and got the same: > > Actually this "ls" output and the last

In addition to my last two messages about Samba updating the password foor root when unix sync is desired: I have found out the cause of this password change thing. With the parm "passwd chat debug = Yes" and "log level = 100" When I look in the logs while changing the password with unix sync, the following happens: [2001/03/27 09:34:20, 10, effective(0, 0), real(0, 0)]

All, I had updated to 1.2.4 right when it came out. I had been working just fine. Today I seem to be having recuring seg faults. can explain it. How can I find why? Anyone else experiencing this? I am running (2) TDM04B cards (has been working since 1.0.9) I have a handfull of UIP200 phones and 1 cisco 7960. I have a unused broadvoic connection that I commented out the registration statement

Thanks to help from Jim Dickenson I managed to start a macro and get info about the channel that picked up the call from my ruby script. The only thing that I cant do so far, is capturing the ${CHANNEL} variable in the ruby script that started the macro. Is that variable accessible from the ruby script too or just from the macro? Here's a snippet from my ruby script:

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

On 12/31/2014 5:05 PM, Robert Blayzor wrote: > On Dec 10, 2014, at 1:52 AM, Steffen Kaiser <skdovecot at smail.inf.fh-brs.de> wrote: > > I've been following this thread and have been seeing a similar problem. Dovecot 2.2.5 and pigeonhole-0.4.6 > > Yet, dovecot still tries to compile it under the user in that path. > > > Dec 31 15:55:11 dovecot: lda(fred): Error:

Dear All, a friend of mine who has a small office want to have a configuration like this: * around 3 people can have full access to internet * around 1-2 people can just have email access (can send and receive email to any address) * the rest of the people in the company just can have internal email (can send and receive email just to/from the peer - on the same domain). I think I can

It has been running flawlessly for quite some time until the update. Global scripts were compiled: /usr/local/etc/dovecot/sieve # ls 10-move-spam.sieve 10-move-spam.svbin However, I ran sievec again and tried saving a modified script and got the same: shiofuki dovecot: lda(gessel at blackrosetech.com): Error: sieve: binary save: failed to create temporary file:

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

Here is another log from the * server CLI, I reall hope some one can help me out on this one. thanks |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time < "" and lead_id != '';| -- VDAD get agent: |0|update of vla table: |127.0.0.1 |UPDATE vicidial_live_agents set

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition?

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

The following strange conditions is happening while I try to dial a SIP user from another SIp user. SIP to Zap dialing is fine, as all 4 users can call PSTN. I'm using Asterisk SVN-branch-1.2-r51359M Example: extension 3210 calls extension 3213. They are all registered properly: chrom01*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 3213/3213

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from