search for: call_logs

*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

Hi there, Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4. here are the CLI actual logs: -- Executing SetAccount("Local/50015308467418@default-ca2e,2", "XXXXXX") in new stack -- Executing AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI

Hello everyone. I need a quick help on how to capture who answered the call with agi. Here is an example: -- Zap/32-1 is ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this:

Hi sir, I was installed Goautodial server and I have b410p BRI card. BRI card showing OK with dahdi_tool, this NT mode. whenever I am dialing from server i am not able to connect the call . in Cli below mention warning is comming . please what is the mistake with me . help me Executing [0559566768 at default:1] AGI("Console/dsp", "agi:// 127.0.0.1:4577/call_log") in new

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to

...back(transfer) ;play transfer (please hold whilst i try to conenct you) ;exten => s,3,Macro(record-start) ; WORKS but starts recording before connect ;exten => s,4,Dial(${ARG2},10,tT)) ; WORKS but starts recording before connect exten => s,3,SetVar(CALLFILENAME=/tmp/call_logs/${ARG1}/${TIMESTAMP}-${CALLERID}) exten => s,4,Dial(${ARG2},10,tTM(record-start)) exten => s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) ;--<SNIP>-- ;-----------------------------------------------------------------------...

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes

Help! I'm (still) having issues with Asterisk Queues. I want to implement a queue so that callers get the 'all our staff are busy at the moment, your call has been placed in a queue and will be answered by the first available operator. You may press 1 at any time to leave a voicemail' announcement, then they can press 1 and leave a voicemail. Documentation with * 1.2.x (I'm

I have a TE210P digium card that has 2 E1/T1 ports. the code in my extensions.conf file for span 1 is : [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This