<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>I have noticed the same on the CLI while calling out Directly, the CLI does not show Ringing event..<br><br><span style="font-weight: bold;"> -- Executing [91XXXXXXXXXX@default:1] Dial("SIP/sanjay-09a0a970", "ZAP/G0/1</span><span style="font-weight: bold;">XXXXXXXXXX</span><span style="font-weight: bold;">")</span><br style="font-weight: bold;"><span style="font-weight: bold;"> -- Called G0/1</span><span style="font-weight: bold;">XXXXXXXXXX</span><br style="font-weight: bold;"><span style="font-weight: bold;"> -- Zap/4-1 answered SIP/sanjay-09a0a970</span><br style="font-weight: bold;"><span style="font-weight: bold;"> -- Hungup 'Zap/4-1'<br></span><br>In the above case, when the CLI prints that <span style="font-weight: bold;">Zap/4-1 answered SIP/sanjay-09a0a970</span> actually call has not yet been picked by anyone, it is still ringing. <br><br><br>Where as one of our other server where we have T1, the CLI looks like below when calling out<br><br> <span style="font-weight: bold;"> -- Executing [91XXXXXXXXXX@internal:1] Dial("SIP/sanjay-09a0a970", "ZAP/G2/1XXXXXXXXXX")</span><br style="font-weight: bold;"> <span style="font-weight: bold;"> -- Called G2/</span><span style="font-weight: bold;">1XXXXXXXXXX</span><br style="font-weight: bold;"><span style="font-weight: bold;"> -- Zap/23-1 is proceeding passing it to SIP/sanjay-08f58048</span><br style="font-weight: bold;"><span style="font-weight: bold;"> -- Zap/23-1 is ringing</span><br style="font-weight: bold;"><span style="font-weight: bold;"> -- Hungup 'Zap/23-1'<br><br></span>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<span style="font-weight: bold;"></span> what can be wrong?<br><br><span style="font-weight: bold;"><br></span>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" <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, New Delhi<br>Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects.<br><br><style>p { margin: 0; }</style><style>body { font-family: 'Times New Roman'; font-size: 12pt; color: #000000}</style>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, New Delhi<br>Subject: [asterisk-users] Call Placed through Manager connecting before the call connects.<br><br><style>p { margin: 0; }</style><style>body { font-family: 'Times New Roman'; font-size: 12pt; color: #000000}</style>Hello,<br><br>I am trying to place call through the Manager, using the Zap Card the call connect to the designated Extension before the call is actually Answered by someone or the Voicemail.<br><br>The message that I am sending is <br><br><span style="font-weight: bold;">Action: Originate</span><br style="font-weight: bold;"><span style="font-weight: bold;">Channel: ZAP/G0/1XXXXXXXXXX</span><br style="font-weight: bold;"><span style="font-weight: bold;">MaxRetries: 0</span><br style="font-weight: bold;"><span style="font-weight: bold;">Context: Test<br></span><span style="font-weight: bold;">Exten: 6563</span><span style="font-weight: bold;"></span><br style="font-weight: bold;"><span style="font-weight: bold;">Priority: 1</span><br style="font-weight: bold;"><span style="font-weight: bold;">CallerID: TEST <1234></span><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 can hear the Ring on the Phone after Bridging.<br>If I try the same for SIP channel I get addition events as <span style="font-weight: bold;">Ringing.</span> <br><br>I want to play a message once the call connects, In this case the message is Played while the phone is Ringing. <br><br>Please help. <br><br><br>Regards,<br>Sanjay Rajdev<br><br>_______________________________________________
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