<html><head><style type='text/css'>p { margin: 0; }</style><style type='text/css'>body { font-family: 'Times New Roman'; font-size: 12pt; color: #000000}</style></head><body>Thanks a lot, will try that out and let you know.<br><br>Regards,<br>Sanjay Rajdev<br><br>----- Original Message -----<br>From: "Nicolás Gudiño" <asternic@gmail.com><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br>Sent: Tuesday, May 27, 2008 12:14:30 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi<br>Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects.<br><br>Hello,<br><br>> Is there no one who can even comment on below?<br>><br><br>Analog zap without callprogress will Answer the line as soon as it<br>starts dialing... You will have to experiment with callprogress,<br>polarity switches, etc.. It was discussed many times. Check<br>zapata.conf for those parameters.<br><br><br>> Regards,<br>> Sanjay Rajdev<br>><br>> ----- Original Message -----<br>> From: "Sanjay Rajdev" <sanjay.rajdev@featherstoneinformatics.com><br>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>> <asterisk-users@lists.digium.com><br>> Sent: Thursday, May 22, 2008 8:47:13 PM GMT +05:30 Chennai, Kolkata, Mumbai,<br>> New Delhi<br>> Subject: Re: [asterisk-users] Call Placed through Manager connecting before<br>> the call connects.<br>><br>> I have noticed the same on the CLI while calling out Directly, the CLI does<br>> not show Ringing event..<br>><br>> -- Executing [91XXXXXXXXXX@default:1] Dial("SIP/sanjay-09a0a970",<br>> "ZAP/G0/1XXXXXXXXXX")<br>> -- Called G0/1XXXXXXXXXX<br>> -- Zap/4-1 answered SIP/sanjay-09a0a970<br>> -- Hungup 'Zap/4-1'<br>><br>> In the above case, when the CLI prints that Zap/4-1 answered<br>> SIP/sanjay-09a0a970 actually call has not yet been picked by anyone, it is<br>> still ringing.<br>><br>><br>> Where as one of our other server where we have T1, the CLI looks like below<br>> when calling out<br>><br>> -- Executing [91XXXXXXXXXX@internal:1] Dial("SIP/sanjay-08f58048",<br>> "ZAP/G2/1XXXXXXXXXX")<br>> -- Called G2/1XXXXXXXXXX<br>> -- Zap/23-1 is proceeding passing it to SIP/sanjay-08f58048<br>> -- Zap/23-1 is ringing<br>> -- Hungup 'Zap/23-1'<br>><br>> This one properly works as it should.<br>><br>> I am not able to find whether this is Asterisk problem or Zaptel problem.<br>><br>> Can someone please suggest what can be wrong?<br>><br>><br>> Regards,<br>> Sanjay Rajdev<br>><br>> ----- Original Message -----<br>> From: "Sanjay Rajdev" <sanjay.rajdev@featherstoneinformatics.com><br>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>> <asterisk-users@lists.digium.com><br>> Cc: "Mailing List Asterisk" <asterisk-users@lists.digium.com><br>> Sent: Thursday, May 22, 2008 7:04:34 PM GMT +05:30 Chennai, Kolkata, Mumbai,<br>> New Delhi<br>> Subject: Re: [asterisk-users] Call Placed through Manager connecting before<br>> the call connects.<br>><br>> I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card.<br>><br>> Regards,<br>> Sanjay Rajdev<br>><br>> ----- Original Message -----<br>> From: "Sanjay Rajdev" <sanjay.rajdev@featherstoneinformatics.com><br>> To: "Mailing List Asterisk" <asterisk-users@lists.digium.com><br>> Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai,<br>> New Delhi<br>> Subject: [asterisk-users] Call Placed through Manager connecting before the<br>> call connects.<br>><br>> Hello,<br>><br>> I am trying to place call through the Manager, using the Zap Card the call<br>> connect to the designated Extension before the call is actually Answered by<br>> someone or the Voicemail.<br>><br>> The message that I am sending is<br>><br>> Action: Originate<br>> Channel: ZAP/G0/1XXXXXXXXXX<br>> MaxRetries: 0<br>> Context: Test<br>> Exten: 6563<br>> Priority: 1<br>> CallerID: TEST <1234><br>><br>><br>> The Events that I get from Manger are<br>> 1. Newchannel<br>> 2. Newcallerid<br>> 3. Newcallerid<br>> 4. Newstate [Here State is changed to Dialing]<br>> 5. Newstate [Here State is changed to Up]<br>> 6. Newexten [Here call is bridged to 6563]<br>><br>> Once the call is Bridged to 6563, the phone is actually not Answered, you<br>> can hear the Ring on the Phone after Bridging.<br>> If I try the same for SIP channel I get addition events as Ringing.<br>><br>> I want to play a message once the call connects, In this case the message is<br>> Played while the phone is Ringing.<br>><br>> Please help.<br>><br>><br>> Regards,<br>> Sanjay Rajdev<br>><br><br><br><br>-- <br>Nicolás Gudiño<br>Buenos Aires - Argentina<br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></body></html>