VoIP Mailing List Archives

[sanjay.rajdev at feath...]

Hello,

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.

The message that I am sending is

Action: Originate
Channel: ZAP/G0/1XXXXXXXXXX
MaxRetries: 0
Context: Test
Exten: 6563
Priority: 1
CallerID: TEST <1234>
The Events that I get from Manger are
1. Newchannel
2. Newcallerid
3. Newcallerid
4. Newstate [Here State is changed to Dialing]
5. Newstate [Here State is changed to Up]
6. Newexten [Here call is bridged to 6563]

Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging.
If I try the same for SIP channel I get addition events as Ringing.

I want to play a message once the call connects, In this case the message is Played while the phone is Ringing.

Please help.


Regards,
Sanjay Rajdev
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[sanjay.rajdev at feath...]

I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card.

Regards,
Sanjay Rajdev

----- Original Message -----
From: "Sanjay Rajdev" <sanjay.rajdev at featherstoneinformatics.com>
To: "Mailing List Asterisk" <asterisk-users at lists.digium.com>
Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: [asterisk-users] Call Placed through Manager connecting before the call connects.

Hello,

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.

The message that I am sending is

Action: Originate
Channel: ZAP/G0/1XXXXXXXXXX
MaxRetries: 0
Context: Test
Exten: 6563
Priority: 1
CallerID: TEST <1234>
The Events that I get from Manger are
1. Newchannel
2. Newcallerid
3. Newcallerid
4. Newstate [Here State is changed to Dialing]
5. Newstate [Here State is changed to Up]
6. Newexten [Here call is bridged to 6563]

Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging.
If I try the same for SIP channel I get addition events as Ringing.

I want to play a message once the call connects, In this case the message is Played while the phone is Ringing.

Please help.


Regards,
Sanjay Rajdev

_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[sanjay.rajdev at feath...]

I have noticed the same on the CLI while calling out Directly, the CLI does not show Ringing event..

-- Executing [91XXXXXXXXXX at default:1] Dial("SIP/sanjay-09a0a970", "ZAP/G0/1 XXXXXXXXXX ")
-- Called G0/1 XXXXXXXXXX
-- Zap/4-1 answered SIP/sanjay-09a0a970
-- Hungup 'Zap/4-1'

In the above case, when the CLI prints that Zap/4-1 answered SIP/sanjay-09a0a970 actually call has not yet been picked by anyone, it is still ringing.
Where as one of our other server where we have T1, the CLI looks like below when calling out

-- Executing [ 91XXXXXXXXXX @internal:1] Dial("SIP/sanjay-09a0a970", "ZAP/G2/1 XXXXXXXXXX ")
-- Called G2/ 1XXXXXXXXXX
-- Zap/23-1 is proceeding passing it to SIP/sanjay-08f58048
-- Zap/23-1 is ringing
-- Hungup 'Zap/23-1'

This one properly works as it should.

I am not able to find whether this is Asterisk problem or Zaptel problem.

Can someone please suggest what can be wrong?


Regards,
Sanjay Rajdev

----- Original Message -----
From: "Sanjay Rajdev" <sanjay.rajdev at featherstoneinformatics.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Cc: "Mailing List Asterisk" <asterisk-users at lists.digium.com>
Sent: Thursday, May 22, 2008 7:04:34 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card.

Regards,
Sanjay Rajdev

----- Original Message -----
From: "Sanjay Rajdev" <sanjay.rajdev at featherstoneinformatics.com>
To: "Mailing List Asterisk" <asterisk-users at lists.digium.com>
Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: [asterisk-users] Call Placed through Manager connecting before the call connects.

Hello,

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.

The message that I am sending is

Action: Originate
Channel: ZAP/G0/1XXXXXXXXXX
MaxRetries: 0
Context: Test
Exten: 6563
Priority: 1
CallerID: TEST <1234>


The Events that I get from Manger are
1. Newchannel
2. Newcallerid
3. Newcallerid
4. Newstate [Here State is changed to Dialing]
5. Newstate [Here State is changed to Up]
6. Newexten [Here call is bridged to 6563]

Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging.
If I try the same for SIP channel I get addition events as Ringing.

I want to play a message once the call connects, In this case the message is Played while the phone is Ringing.

Please help.


Regards,
Sanjay Rajdev

_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[sanjay.rajdev at feath...]

Is there no one who can even comment on below?

Regards,
Sanjay Rajdev

----- Original Message -----
From: "Sanjay Rajdev" <sanjay.rajdev at featherstoneinformatics.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Sent: Thursday, May 22, 2008 8:47:13 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

I have noticed the same on the CLI while calling out Directly, the CLI does not show Ringing event..

-- Executing [91XXXXXXXXXX at default:1] Dial("SIP/sanjay-09a0a970", "ZAP/G0/1 XXXXXXXXXX ")
-- Called G0/1 XXXXXXXXXX
-- Zap/4-1 answered SIP/sanjay-09a0a970
-- Hungup 'Zap/4-1'

In the above case, when the CLI prints that Zap/4-1 answered SIP/sanjay-09a0a970 actually call has not yet been picked by anyone, it is still ringing.
Where as one of our other server where we have T1, the CLI looks like below when calling out

-- Executing [91XXXXXXXXXX at internal:1] Dial("SIP/sanjay- 08f58048 ", "ZAP/G2/1XXXXXXXXXX")
-- Called G2/ 1XXXXXXXXXX
-- Zap/23-1 is proceeding passing it to SIP/sanjay-08f58048
-- Zap/23-1 is ringing
-- Hungup 'Zap/23-1'

This one properly works as it should.

I am not able to find whether this is Asterisk problem or Zaptel problem.

Can someone please suggest what can be wrong?


Regards,
Sanjay Rajdev

----- Original Message -----
From: "Sanjay Rajdev" <sanjay.rajdev at featherstoneinformatics.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Cc: "Mailing List Asterisk" <asterisk-users at lists.digium.com>
Sent: Thursday, May 22, 2008 7:04:34 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card.

Regards,
Sanjay Rajdev

----- Original Message -----
From: "Sanjay Rajdev" <sanjay.rajdev at featherstoneinformatics.com>
To: "Mailing List Asterisk" <asterisk-users at lists.digium.com>
Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: [asterisk-users] Call Placed through Manager connecting before the call connects.

Hello,

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.

The message that I am sending is

Action: Originate
Channel: ZAP/G0/1XXXXXXXXXX
MaxRetries: 0
Context: Test
Exten: 6563
Priority: 1
CallerID: TEST <1234>


The Events that I get from Manger are
1. Newchannel
2. Newcallerid
3. Newcallerid
4. Newstate [Here State is changed to Dialing]
5. Newstate [Here State is changed to Up]
6. Newexten [Here call is bridged to 6563]

Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging.
If I try the same for SIP channel I get addition events as Ringing.

I want to play a message once the call connects, In this case the message is Played while the phone is Ringing.

Please help.


Regards,
Sanjay Rajdev

_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[abalashov at evaristes...]

Sanjay Rajdev wrote:

[asternic at gmail.com]

Hello,

[sanjay.rajdev at feath...]

Thanks a lot, will try that out and let you know.

Regards,
Sanjay Rajdev

----- Original Message -----
From: "Nicol?s Gudi?o" <asternic at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Sent: Tuesday, May 27, 2008 12:14:30 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi
Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

Hello,

[eric at fnords.org]

callprogress=yes is commonly known in the Asterisk community as
"randomlydisconnectmycalls=yes". It does not work well. The .sample
config file even says it is experimental. If you can accept some
percentage of calls being randomly disconnected, then by all means use
the option.

Sanjay Rajdev wrote: