Re: [asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..

2010-07-09 Thread Massimo Nuvoli
Zeeshan Zakaria ha scritto:
 I have two test asterisk boxes, both version 1.4.26, on which I do
 Answer() followed by MusicOnHold() and it works just fine. I do this all
 the time as this is my standard way of testing new contexts.

Yesterday i tested another installation and i found the same issue.

Maybe the problem is SIP related or console channel related.

I explain (if someone can do a test i am happy).

Go to the asterisk console, place a dial command calling thru the
SIP trunk, then place a transfer to the extension MusicOnHold after
the Answer...

(this is the sequence)

dial 0num...@from-sip (the from-sip is the context where a sip phone
can dial to the trunk)
pick up the phone called
transfer *...@from-sip (the *199 extension is Answer - MusicOnHold)
you must hear the music on the phone called (or not)

So this may be a console channel problem...

Yesterday i try to use the outgoing spool (place a file on
/var/spool/asterisk/outgoing making a call to the phone and directly
go to the *199 extension, the same thing i do on console automated
with no console channel), audio ok.

So i am going to open a bug... :-)

Thnks.

 Zeeshan A Zakaria
 
 --
 www.ilovetovoip.com http://www.ilovetovoip.com
 
 On 2010-07-07 4:16 AM, Massimo Nuvoli mass...@archivio.it
 mailto:mass...@archivio.it wrote:

 I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?

 I spend 4 hours to try to solve... but found only a workaround.

 As is easy to reproduce the problem i need to know if this is a bug or
 if there is some idiot configuration that i miss.

 Maybe also the bug is know...

 Scenario:

 Asterisk installation on ubuntu 9.04 64 bit.

 Trunk SIP (two different providers)

 On the Asterisk server there are a number of SIP clients.

 If i use the sip client all things ok, i made a call and everything ok.

 If i place the call from the server (or if i call trhu the SIP trunk
 the asterisk system) everytime the Answer() application seeems to NOT
 work.

 The only way to make it work is to use some other function that do the
 Answer in place.

 (call coming from the SIP trunk)
 If i use

 Answer()
 MusicOnHold()

 I hear nothing.

 If i use

 Answer()
 PlayBack(silence/1)
 MusicOnHold()

 or

 Answer()
 VoiceMail(1...@default)

 i can hear all ok (it seems that the PlayBack and the VoiceMail apps
 are able to Answer really...)

 I checked the SIP debug trace, it seems no problem on the SIP side.

 Thnks guys.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 

attachment: massimo.vcf

signature.asc
Description: OpenPGP digital signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..

2010-07-07 Thread Zeeshan Zakaria
Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-07-07 4:16 AM, Massimo Nuvoli mass...@archivio.it wrote:

I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?

I spend 4 hours to try to solve... but found only a workaround.

As is easy to reproduce the problem i need to know if this is a bug or
if there is some idiot configuration that i miss.

Maybe also the bug is know...

Scenario:

Asterisk installation on ubuntu 9.04 64 bit.

Trunk SIP (two different providers)

On the Asterisk server there are a number of SIP clients.

If i use the sip client all things ok, i made a call and everything ok.

If i place the call from the server (or if i call trhu the SIP trunk
the asterisk system) everytime the Answer() application seeems to NOT
work.

The only way to make it work is to use some other function that do the
Answer in place.

(call coming from the SIP trunk)
If i use

Answer()
MusicOnHold()

I hear nothing.

If i use

Answer()
PlayBack(silence/1)
MusicOnHold()

or

Answer()
VoiceMail(1...@default)

i can hear all ok (it seems that the PlayBack and the VoiceMail apps
are able to Answer really...)

I checked the SIP debug trace, it seems no problem on the SIP side.

Thnks guys.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..

2010-07-07 Thread Zeeshan Zakaria
I have two test asterisk boxes, both version 1.4.26, on which I do Answer()
followed by MusicOnHold() and it works just fine. I do this all the time as
this is my standard way of testing new contexts.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-07-07 4:16 AM, Massimo Nuvoli mass...@archivio.it wrote:

I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?

I spend 4 hours to try to solve... but found only a workaround.

As is easy to reproduce the problem i need to know if this is a bug or
if there is some idiot configuration that i miss.

Maybe also the bug is know...

Scenario:

Asterisk installation on ubuntu 9.04 64 bit.

Trunk SIP (two different providers)

On the Asterisk server there are a number of SIP clients.

If i use the sip client all things ok, i made a call and everything ok.

If i place the call from the server (or if i call trhu the SIP trunk
the asterisk system) everytime the Answer() application seeems to NOT
work.

The only way to make it work is to use some other function that do the
Answer in place.

(call coming from the SIP trunk)
If i use

Answer()
MusicOnHold()

I hear nothing.

If i use

Answer()
PlayBack(silence/1)
MusicOnHold()

or

Answer()
VoiceMail(1...@default)

i can hear all ok (it seems that the PlayBack and the VoiceMail apps
are able to Answer really...)

I checked the SIP debug trace, it seems no problem on the SIP side.

Thnks guys.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users