[asterisk-users] New Release asterisk 1.6 Beta

[Naveen Palani]

Hi,

Was just wondering about the bug fix in asterisk-1.6 Beta Release..

Its been said that "Added an 'n' option to SpeechBackground to request that the 
channel not get answered".

Can anyone brief me about what exactly this bug are they talking about. We had 
issues with Background syntax working in  macro. Is this the bug fix they are 
talking about.

Regards,
Naveen.Palani
Quinnox Consultancy Services Ltd | Pune | INDIA |
Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 |
Fax : +91 20 4015 2305 | Email : [EMAIL PROTECTED]



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[asterisk-users] SIP Registration!

[Naveen Palani]

Hi,

I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel 
sip providers.

Use DID numbers for the incoming calls which works fine when i dont use any 
peer setting in my sip.conf file. But when i use a peer and make calls thru the 
DID number it doesn't reach asterisk at all. Doesnt give me any errors as well.

peer in my sip conf is as given below:

[proxy2_bandtel]
type=peer
username=206**1
secret=***
fromdomain=206**1
host=proxy1.bandtel.com
qualify=yes
outboundproxy=proxy1.bandtel.com

If i dont use the above code in sip.conf file the DID number reaches asterisk 
and completes the incoming call. But i need to have the above code to register 
SIP, which should give me the status as ok when i run the command 'sip show 
peers'.

Can you please let me know why this doesnt work with the above code.

Thanks and appreciate your response.

Regards,
Naveen.Palani


“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
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[asterisk-users] Gtalk with asterisk

[Naveen Palani]

Hi,

I have been working with Asterisk for the ivr functionalities in the past. I am 
interested to implement the Jabber - Gtalk in asterisk.

For which i installed the iksemel but this didnt help me out. I couldnt find 
the res_jabber.so file any where in the asterisk source directory. Still when i 
run the command "make menuselect" the channel driver "chan_gtalk" shows xxx 
(dependencies not met). How can i register gtalk with asterisk.

If you can provide me with some basic details i can carry forward.

Thanks and appreciate your response.

Regards,
Naveen.Palani



“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
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Re: [asterisk-users] No compatible codecs!

[Naveen Palani]

I read from the forums, that if i build mysql the problem will be resolved. As 
i get the similar warning message for MeetMe().

How to build mysql or MeetMe manually??

Regards,
Naveen.Palani


- Original Message -
From: Naveen Palani<mailto:[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Sent: Tuesday, February 19, 2008 3:17 PM
Subject: Re: No compatible codecs!

Resolved!

The problem was with sip.conf file. I had to comment the lines

allow=alaw
allow=ulaw

This made the trick..


I am trying to get the mysql database connection from my asterisk box. 
Installed the asterisk-addons-1.4.4 version on the same box where asterisk and 
mysql is installed.

When i use the following line in dialplan:

exten => 800,2,MYSQL(Connect connid localhost root newpwd asterisk)

Gives me the warning error message:

[Feb 19 16:35:34] WARNING[10187]: pbx.c:1797 pbx_extension_helper: No 
application 'MYSQL' for extension (default, 800, 2)
  == Spawn extension (default, 800, 2) exited non-zero on 'SIP/102-081a6dc0'

Can some one let me know what iam i missing here.

Thanks and appreciate your response.

Regards,
Naveen.Palani


- Original Message -
From: Naveen Palani<mailto:[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Sent: Tuesday, February 19, 2008 12:23 PM
Subject: No compatible codecs!

Hi,

I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try 
making a simple incoming call using xlite softphone. I get the following 
message when i try calling to the number.

*CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No 
compatible codecs, not accepting this offer!

Which codec is it talking abt here. How can i resolve this.

My dialplan is as given below:

extensions.conf:

exten => 800,1,Answer()
exten => 800,2,Playback(vm-saved)
exten => 800,3,Hangup

Regards,
Naveen.Palani


“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
Delivery Model.”

This e-mail and any attached files are confidential, proprietary, and may also 
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and delete the e-mail and any attached files and destroy any copies of it; you 
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Quinnox Consultancy Services and/or any of its sister companies owns no 
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Services.

Unauthorized reading, reproduction, publication, use, dissemination, 
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Re: [asterisk-users] No compatible codecs!

[Naveen Palani]

Resolved!

The problem was with sip.conf file. I had to comment the lines

allow=alaw
allow=ulaw

This made the trick..


I am trying to get the mysql database connection from my asterisk box. 
Installed the asterisk-addons-1.4.4 version on the same box where asterisk and 
mysql is installed.

When i use the following line in dialplan:

exten => 800,2,MYSQL(Connect connid localhost root newpwd asterisk)

Gives me the warning error message:

[Feb 19 16:35:34] WARNING[10187]: pbx.c:1797 pbx_extension_helper: No 
application 'MYSQL' for extension (default, 800, 2)
  == Spawn extension (default, 800, 2) exited non-zero on 'SIP/102-081a6dc0'

Can some one let me know what iam i missing here.

Thanks and appreciate your response.

Regards,
Naveen.Palani


- Original Message -----
From: Naveen Palani<mailto:[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Sent: Tuesday, February 19, 2008 12:23 PM
Subject: No compatible codecs!

Hi,

I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try 
making a simple incoming call using xlite softphone. I get the following 
message when i try calling to the number.

*CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No 
compatible codecs, not accepting this offer!

Which codec is it talking abt here. How can i resolve this.

My dialplan is as given below:

extensions.conf:

exten => 800,1,Answer()
exten => 800,2,Playback(vm-saved)
exten => 800,3,Hangup

Regards,
Naveen.Palani


“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
Delivery Model.”

This e-mail and any attached files are confidential, proprietary, and may also 
be legally privileged information, and are intended solely for the use of the 
individual or entity to whom they are addressed. If you are not the intended 
recipient of this e-mail, please send it back to the person who sent it to you 
and delete the e-mail and any attached files and destroy any copies of it; you 
may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED]

Quinnox Consultancy Services and/or any of its sister companies owns no 
responsibility for the views presented in the e-mail and any attached files 
unless the sender mentions so, with due authority of Quinnox Consultancy 
Services.

Unauthorized reading, reproduction, publication, use, dissemination, 
forwarding, printing or copying of this e-mail and its attachments is 
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liability, in case of any damage caused by a non-detected virus.

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[asterisk-users] No compatible codecs!

[Naveen Palani]

Hi,

I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try 
making a simple incoming call using xlite softphone. I get the following 
message when i try calling to the number.

*CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No 
compatible codecs, not accepting this offer!

Which codec is it talking abt here. How can i resolve this.

My dialplan is as given below:

extensions.conf:

exten => 800,1,Answer()
exten => 800,2,Playback(vm-saved)
exten => 800,3,Hangup

Regards,
Naveen.Palani


“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
Delivery Model.”

This e-mail and any attached files are confidential, proprietary, and may also 
be legally privileged information, and are intended solely for the use of the 
individual or entity to whom they are addressed. If you are not the intended 
recipient of this e-mail, please send it back to the person who sent it to you 
and delete the e-mail and any attached files and destroy any copies of it; you 
may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED]

Quinnox Consultancy Services and/or any of its sister companies owns no 
responsibility for the views presented in the e-mail and any attached files 
unless the sender mentions so, with due authority of Quinnox Consultancy 
Services.

Unauthorized reading, reproduction, publication, use, dissemination, 
forwarding, printing or copying of this e-mail and its attachments is 
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We have checked this message for any known viruses; however we decline any 
liability, in case of any damage caused by a non-detected virus.

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[asterisk-users] Pass arguments from extensions.conf

[Naveen Palani]

Hi,

I have been working with asterisk to make ivr calls (outbound and inbound). I 
have the functionality -

Read(variable|file_name)

used in my dialplan. Now i need to pass the variable to my ruby file to compare 
the data entered with the database (mysql).

How can i pass the arguments from my dialplan to the ruby file. Is there a way 
i can do it with the agi script?

Any one has any clues on it.

Regards,
Naveen.Palani



“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
Delivery Model.”

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individual or entity to whom they are addressed. If you are not the intended 
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and delete the e-mail and any attached files and destroy any copies of it; you 
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Quinnox Consultancy Services and/or any of its sister companies owns no 
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Unauthorized reading, reproduction, publication, use, dissemination, 
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[asterisk-users] asterisk to mysql database!

[Naveen Palani]

Hello,

Is there a possibility to connect from asterisk to mysql database without the 
interface application like Ruby or PHP.

If i can connect to mysql database from asterisk, i can update the database for 
manipulations.

Appreciate your response.

Regards,
Naveen.Palani


“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
Delivery Model.”

This e-mail and any attached files are confidential, proprietary, and may also 
be legally privileged information, and are intended solely for the use of the 
individual or entity to whom they are addressed. If you are not the intended 
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and delete the e-mail and any attached files and destroy any copies of it; you 
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Re: [asterisk-users] Interrupt the swift text

[Naveen Palani]

Steve/Ron,

I also did that. I created a wave file and stored in /tmp directory and then
use Background cmd inside macro. But it doesnt seem to work.

I saw from the forum to use Background with the context parameter set to the
macro context."

Used it in this way, suggest me if it is wrong:

exten => s,3,Background(/tmp/test|outbound-connect)

But still doesnt work. Please suggest me.

Regards,
Naveen


- Original Message -
From: "Ron Joffe" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, January 15, 2008 12:33 PM
Subject: Re: [asterisk-users] Interrupt the swift text


On Tuesday 15 January 2008 12:32, Naveen Palani wrote:
> Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000
> "Press 1 to confirm. Press 3 to cancel."

Naveen,

How about generating the wav files and storing them, then playing the wav's
from the call tree, rather then re-generating the TTS every time. These seem
to be static in nature.

Ron


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E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
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Delivery Model.”

This e-mail and any attached files are confidential, proprietary, and may also 
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individual or entity to whom they are addressed. If you are not the intended 
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and delete the e-mail and any attached files and destroy any copies of it; you 
may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED]

Quinnox Consultancy Services and/or any of its sister companies owns no 
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Unauthorized reading, reproduction, publication, use, dissemination, 
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[asterisk-users] Interrupt the swift text

[Naveen Palani]

Hi,

I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift 
text to audio.

My functionality is as for example i make this text to audio deliver the person 
called.

Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 
1 to confirm. Press 3 to cancel."

extension.conf dialplan:

[dialout]
exten => 
outbound-handler,1,Dial(SIP/102,60,gM(outbound-connect^agi://10.1.1.68/ivr/speak^${CallInitiate_hashdata}))

[macro-outbound-connect]
exten => s,1,Answer()
exten => s,2,System(swift -o /tmp/test.wav -p 
audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to 
cancel.")
exten => s,3,Background(/tmp/test)
exten => s,4,Hangup

exten => 1,1,Playback(thanks)
exten => 2,1,Playback(bye)

Here in this, the call is connected and answered the control transfer to macro 
context. One way i can interrupt the text before it completes the text is by 
using 'Background (/tmp/test)' to play the audio.

When iam in the middle of the audio if i press 1 before it completes the entire 
text, the control should go to 'exten => 1,1,Playback(thanks)'. But in macro 
the 'Background' doesnt seem to work. It works fine outside macro context.

When i use the Asterisk cmd GoTo(new_context,extn,priority) inside macro, I get 
a message 'channel jumping out of macro "outbound-connect"' waits for a minute 
and hungs up, the control doesnt go to new_context.

Does anyone have any ideas i can work it out. How can i have the Asterisk cmd 
Background inside macro? or how to execute the GoTo command?

Thanks and appreciate your response.

Regards,
Naveen.Palani



“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
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This e-mail and any attached files are confidential, proprietary, and may also 
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individual or entity to whom they are addressed. If you are not the intended 
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and delete the e-mail and any attached files and destroy any copies of it; you 
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responsibility for the views presented in the e-mail and any attached files 
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[asterisk-users] Fax machine detect

[Naveen Palani]

Hi,

I was recently trying out with AMD (Answering Machine Detect) to detect the 
status of my call if it being picked up by HUMAN or MACHINE.

Just want to know if any supporting features in asterisk 1.4.11 to detect if 
the call enters the Fax machine.

Please provide the documentation link if any one has ideas on the same.

Appreciate your response.

Regards,
Naveen.Palani
Quinnox Consultancy Services Ltd | Pune | INDIA |
Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 |
Fax : +91 20 4015 2305 | Email : [EMAIL PROTECTED]



“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
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This e-mail and any attached files are confidential, proprietary, and may also 
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individual or entity to whom they are addressed. If you are not the intended 
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and delete the e-mail and any attached files and destroy any copies of it; you 
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Quinnox Consultancy Services and/or any of its sister companies owns no 
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[asterisk-users] Fax machine detect

[Naveen Palani]

Hi,

I was recently trying out with AMD (Answering Machine Detect) to detect the 
status of my call if it being picked up by HUMAN or MACHINE.

Just want to know if any supporting features in asterisk 1.4.11 to detect if 
the call enters the Fax machine.

Please provide the documentation link if any one has ideas on the same.

Appreciate your response.

Regards,
Naveen.Palani
Quinnox Consultancy Services Ltd | Pune | INDIA |
Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 |
Fax : +91 20 4015 2305 | Email : [EMAIL PROTECTED]



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[asterisk-users] g729 codec - simultaneous calls

[Naveen Palani]

Hi,

I use asterisk 1.4.11 version for making outbound calls. Running it on 
linux(fedora core 7) machine. Recently purchased the g729 codec, got it 
registered with my asterisk box. I have two queries for you to help me.

1. How do i know when an outbound call is placed that it makes use of the g729 
codec.

when i use the command "show g729" i get the following:

0/0 encoders/decoders of 4 licensed channels are currently in use

2. When i have g729 codec installed on my asterisk box, i cannot make 
simultaneous calls which i can do if i uninstall it. How can i make 
simultaneous calls with the codec installed?


Thanks and appreciate your response.

Regards,
Naveen.Palani
Quinnox Consultancy Services Ltd | Pune | INDIA |
Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 |
Fax : +91 20 4015 2305 | Email : [EMAIL PROTECTED]



“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, 
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of 
E-Business, ERP, Application Management Services, and EAI to customers in BFSI, 
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global 
Delivery Model.”

This e-mail and any attached files are confidential, proprietary, and may also 
be legally privileged information, and are intended solely for the use of the 
individual or entity to whom they are addressed. If you are not the intended 
recipient of this e-mail, please send it back to the person who sent it to you 
and delete the e-mail and any attached files and destroy any copies of it; you 
may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED]

Quinnox Consultancy Services and/or any of its sister companies owns no 
responsibility for the views presented in the e-mail and any attached files 
unless the sender mentions so, with due authority of Quinnox Consultancy 
Services.

Unauthorized reading, reproduction, publication, use, dissemination, 
forwarding, printing or copying of this e-mail and its attachments is 
prohibited.
We have checked this message for any known viruses; however we decline any 
liability, in case of any damage caused by a non-detected virus.

For more details about our company, visit http://www.Quinnox.com
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