Re: [asterisk-users] Delay before audio starts

2013-03-21 Thread Gerard 
On 03/21/13 14:14, Gerard wrote:
 I think a simple tcpdump of the traffic will show the mystery. It can
 be your provider doing something nasty. Have you tried using some
 other cheap SIP termination? or arrange a fake termination yourself
 on another server?

 Leandro
 
 I thought so too, but it doesn't appear to .
 
 I just bought a door intercom device, set up the extension for it and
 it's doing the same thing, when it connects there is a 10 second delay
 before the other side can hear my voice.
 However watching tcpdump, the audio starts streaming both ways immediately.
 Changing the dialplan fixes the issue:
 957 = { // Test door phone
 Answer(); //  --- this line fixes the problem!
 Dial(SIP/199,20);
 Hangup();
 };
 
 It's an ok workaround for the door intercom, but in the case of the
 forwarded calls below, as soon as I Answer() their ringback disappears
 and the line goes dead while they wait for our guy to answer the phone.
 
 I may start a separate post about getting ringback to work after Answer();

As a followup, hold music instead of ringback works fine, so as my
current workaround, I'm using an mp3 of the ringback sound as the hold
music.
Anything is better then a dead line :)


 
 Thanks for the help by the way.
 -Gerard
 
 
 On 03/01/13 14:34, Leandro Dardini wrote:
 

 2013/3/1 Gerard gsara...@rarcoa.com

 I thought it was the re-invites too, but I have it turned off
 everywhere.

 On 03/01/13 08:36, Eric Wieling wrote:
 When Answer fixes the issue, the root cause is often NAT (could
 be
 firewall) since Answering the call prevents any reinvites.

 -Original Message- From:
 asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
 Sent: Friday, March 01, 2013 9:33 AM To:
 asterisk-users@lists.digium.com Subject: Re: [asterisk-users]
 Delay before audio starts

 I've found a workaround of sorts, If I change my below code to : 
 1AA = { NoOp(${CALLERID(num)}); Answer();  //
 --- add this Ringing; 
 Set(CHANNEL(musicclass)=none); 
 Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };

 That fixes the issue. It doesn't fix the call forward issue on
 the phone
 though. I've made a few extra extensions, one each corresponding to
 a number he wants to call forward to, if I have him forward to the
 extensions who then forward to the real number, it works, thanks to
 adding Answer() to the dialplan.

 -Gerard


 On 02/26/13 13:19, Gerard wrote:
 Hi everyone,

 I'm having a hard time figuring this issue out, we just
 switched from a T1 PRI to a SIP trunk provider and that's when
 the issue started. Now when someone forwards all calls on their
 phone to a cellphone, when a customer calls in, Asterisk
 correctly calls the cellphone and connects the call, but there
 is a long delay before the audio starts, basically for the
 first 6-10 seconds of the call there is dead silence,
 eventually the audio will start and everything works
 correctly. We never had this problem with the PRI. So I suspect
 it has something to do with a call coming in as SIP and going
 out as SIP.

 At first I thought it was a call forwarding issue because I got
 this message in the console: [Feb 26 12:35:19]
 NOTICE[1143][C-025d]: app_dial.c:958 do_forward: Not
 accepting call completion offers from call-forward recipient 
 Local/1XX@default-0013;1

 So I put this in my dial plan:

 1AA = { NoOp(${CALLERID(num)}); Ringing; 
 Set(CHANNEL(musicclass)=none); 
 Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };

 So basically as soon as someone calls incoming number
 AA, Asterisk dials phone number XX. it's a
 quick and dirty way to call forward.. and this does the same
 thing, there's a good 8 second delay before the audio kicks
 in.


 There is a Linux firewall with NAT in the path, but I have no
 other audio issues, so don't *think* it's a factor. I just
 upgraded to asterisk 11.2.1.


 Asterisk 11.2.1 built by root @ phonesys2 on a i686 running
 Linux on 2013-02-23 01:40:02 UTC


 Any help would be appreciated, Thanks,


 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

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Re: [asterisk-users] Delay before audio starts

2013-03-21 Thread Asghar Mohammad 
hi,
exten 000,1.Progress() work in some situation.

On Thu, Mar 21, 2013 at 9:30 PM, Gerard gsara...@rarcoa.com wrote:

 On 03/21/13 14:14, Gerard wrote:
  I think a simple tcpdump of the traffic will show the mystery. It can
  be your provider doing something nasty. Have you tried using some
  other cheap SIP termination? or arrange a fake termination yourself
  on another server?
 
  Leandro
 
  I thought so too, but it doesn't appear to .
 
  I just bought a door intercom device, set up the extension for it and
  it's doing the same thing, when it connects there is a 10 second delay
  before the other side can hear my voice.
  However watching tcpdump, the audio starts streaming both ways
 immediately.
  Changing the dialplan fixes the issue:
  957 = { // Test door phone
  Answer(); //  --- this line fixes the problem!
  Dial(SIP/199,20);
  Hangup();
  };
 
  It's an ok workaround for the door intercom, but in the case of the
  forwarded calls below, as soon as I Answer() their ringback disappears
  and the line goes dead while they wait for our guy to answer the phone.
 
  I may start a separate post about getting ringback to work after
 Answer();

 As a followup, hold music instead of ringback works fine, so as my
 current workaround, I'm using an mp3 of the ringback sound as the hold
 music.
 Anything is better then a dead line :)


 
  Thanks for the help by the way.
  -Gerard
 
 
  On 03/01/13 14:34, Leandro Dardini wrote:
 
 
  2013/3/1 Gerard gsara...@rarcoa.com
 
  I thought it was the re-invites too, but I have it turned off
  everywhere.
 
  On 03/01/13 08:36, Eric Wieling wrote:
  When Answer fixes the issue, the root cause is often NAT (could
  be
  firewall) since Answering the call prevents any reinvites.
 
  -Original Message- From:
  asterisk-users-boun...@lists.digium.com [mailto:
  asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
  Sent: Friday, March 01, 2013 9:33 AM To:
  asterisk-users@lists.digium.com Subject: Re: [asterisk-users]
  Delay before audio starts
 
  I've found a workaround of sorts, If I change my below code to :
  1AA = { NoOp(${CALLERID(num)}); Answer();  //
  --- add this Ringing;
  Set(CHANNEL(musicclass)=none);
  Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };
 
  That fixes the issue. It doesn't fix the call forward issue on
  the phone
  though. I've made a few extra extensions, one each corresponding to
  a number he wants to call forward to, if I have him forward to the
  extensions who then forward to the real number, it works, thanks to
  adding Answer() to the dialplan.
 
  -Gerard
 
 
  On 02/26/13 13:19, Gerard wrote:
  Hi everyone,
 
  I'm having a hard time figuring this issue out, we just
  switched from a T1 PRI to a SIP trunk provider and that's when
  the issue started. Now when someone forwards all calls on their
  phone to a cellphone, when a customer calls in, Asterisk
  correctly calls the cellphone and connects the call, but there
  is a long delay before the audio starts, basically for the
  first 6-10 seconds of the call there is dead silence,
  eventually the audio will start and everything works
  correctly. We never had this problem with the PRI. So I suspect
  it has something to do with a call coming in as SIP and going
  out as SIP.
 
  At first I thought it was a call forwarding issue because I got
  this message in the console: [Feb 26 12:35:19]
  NOTICE[1143][C-025d]: app_dial.c:958 do_forward: Not
  accepting call completion offers from call-forward recipient
  Local/1XX@default-0013;1
 
  So I put this in my dial plan:
 
  1AA = { NoOp(${CALLERID(num)}); Ringing;
  Set(CHANNEL(musicclass)=none);
  Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };
 
  So basically as soon as someone calls incoming number
  AA, Asterisk dials phone number XX. it's a
  quick and dirty way to call forward.. and this does the same
  thing, there's a good 8 second delay before the audio kicks
  in.
 
 
  There is a Linux firewall with NAT in the path, but I have no
  other audio issues, so don't *think* it's a factor. I just
  upgraded to asterisk 11.2.1.
 
 
  Asterisk 11.2.1 built by root @ phonesys2 on a i686 running
  Linux on 2013-02-23 01:40:02 UTC
 
 
  Any help would be appreciated, Thanks,
 
 
 
  --
  _
  -- 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
 


 --
 Gerard Saraber
 Network Admin.
 Rarcoa, Inc
 (630) 654-2580 x199
 (630) 654-3556 (fax)
 (630) 915-4122 (cell

Re: [asterisk-users] Delay before audio starts

2013-03-01 Thread Gerard 
I've found a workaround of sorts, If I change my below code to :
 1AA = {
 NoOp(${CALLERID(num)});
 Answer();  // --- add this
 Ringing;
 Set(CHANNEL(musicclass)=none);
 Dial(${OUTBOUND-TRUNKR}/1XX,30);
 Voicemail(198,u);
  };

That fixes the issue. It doesn't fix the call forward issue on the phone
though. I've made a few extra extensions, one each corresponding to a
number he wants to call forward to, if I have him forward to the
extensions who then forward to the real number, it works, thanks to
adding Answer() to the dialplan.

-Gerard


On 02/26/13 13:19, Gerard wrote:
 Hi everyone,
 
 I'm having a hard time figuring this issue out, we just switched from a
 T1 PRI to a SIP trunk provider and that's when the issue started.
 Now when someone forwards all calls on their phone to a cellphone, when
 a customer calls in, Asterisk correctly calls the cellphone and connects
 the call, but there is a long delay before the audio starts, basically
 for the first 6-10 seconds of the call there is dead silence, eventually
 the audio will start and everything works correctly.
 We never had this problem with the PRI. So I suspect it has something to
 do with a call coming in as SIP and going out as SIP.
 
 At first I thought it was a call forwarding issue because I got this
 message in the console:
 [Feb 26 12:35:19] NOTICE[1143][C-025d]: app_dial.c:958 do_forward:
 Not accepting call completion offers from call-forward recipient
 Local/1XX@default-0013;1
 
 So I put this in my dial plan:
 
 1AA = {
 NoOp(${CALLERID(num)});
 Ringing;
 Set(CHANNEL(musicclass)=none);
 Dial(${OUTBOUND-TRUNKR}/1XX,30);
 Voicemail(198,u);
  };
 
 So basically as soon as someone calls incoming number AA,
 Asterisk dials phone number XX. it's a quick and dirty way to
 call forward.. and this does the same thing, there's a good 8 second
 delay before the audio kicks in.
 
 
 There is a Linux firewall with NAT in the path, but I have no other
 audio issues, so don't *think* it's a factor.
 I just upgraded to asterisk 11.2.1.
 
 
 Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
 2013-02-23 01:40:02 UTC
 
 
 Any help would be appreciated,
 Thanks,
 


--
_
-- 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] Delay before audio starts

2013-03-01 Thread Eric Wieling 
When Answer fixes the issue, the root cause is often NAT (could be firewall) 
since Answering the call prevents any reinvites.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Friday, March 01, 2013 9:33 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Delay before audio starts

I've found a workaround of sorts, If I change my below code to :
 1AA = {
 NoOp(${CALLERID(num)});
 Answer();  // --- add this
 Ringing;
 Set(CHANNEL(musicclass)=none);
 Dial(${OUTBOUND-TRUNKR}/1XX,30);
 Voicemail(198,u);
  };

That fixes the issue. It doesn't fix the call forward issue on the phone 
though. I've made a few extra extensions, one each corresponding to a number he 
wants to call forward to, if I have him forward to the extensions who then 
forward to the real number, it works, thanks to adding Answer() to the 
dialplan.

-Gerard


On 02/26/13 13:19, Gerard wrote:
 Hi everyone,
 
 I'm having a hard time figuring this issue out, we just switched from 
 a
 T1 PRI to a SIP trunk provider and that's when the issue started.
 Now when someone forwards all calls on their phone to a cellphone, 
 when a customer calls in, Asterisk correctly calls the cellphone and 
 connects the call, but there is a long delay before the audio starts, 
 basically for the first 6-10 seconds of the call there is dead 
 silence, eventually the audio will start and everything works correctly.
 We never had this problem with the PRI. So I suspect it has something 
 to do with a call coming in as SIP and going out as SIP.
 
 At first I thought it was a call forwarding issue because I got this 
 message in the console:
 [Feb 26 12:35:19] NOTICE[1143][C-025d]: app_dial.c:958 do_forward:
 Not accepting call completion offers from call-forward recipient
 Local/1XX@default-0013;1
 
 So I put this in my dial plan:
 
 1AA = {
 NoOp(${CALLERID(num)});
 Ringing;
 Set(CHANNEL(musicclass)=none);
 Dial(${OUTBOUND-TRUNKR}/1XX,30);
 Voicemail(198,u);
  };
 
 So basically as soon as someone calls incoming number AA, 
 Asterisk dials phone number XX. it's a quick and dirty way to 
 call forward.. and this does the same thing, there's a good 8 second 
 delay before the audio kicks in.
 
 
 There is a Linux firewall with NAT in the path, but I have no other 
 audio issues, so don't *think* it's a factor.
 I just upgraded to asterisk 11.2.1.
 
 
 Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
 2013-02-23 01:40:02 UTC
 
 
 Any help would be appreciated,
 Thanks,
 


--
_
-- 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

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

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_
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Re: [asterisk-users] Delay before audio starts

2013-03-01 Thread Gerard 
I thought it was the re-invites too, but I have it turned off everywhere.

On 03/01/13 08:36, Eric Wieling wrote:
 When Answer fixes the issue, the root cause is often NAT (could be firewall) 
 since Answering the call prevents any reinvites.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
 Sent: Friday, March 01, 2013 9:33 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Delay before audio starts
 
 I've found a workaround of sorts, If I change my below code to :
  1AA = {
  NoOp(${CALLERID(num)});
Answer();  // --- add this
  Ringing;
  Set(CHANNEL(musicclass)=none);
  Dial(${OUTBOUND-TRUNKR}/1XX,30);
  Voicemail(198,u);
   };
 
 That fixes the issue. It doesn't fix the call forward issue on the phone 
 though. I've made a few extra extensions, one each corresponding to a number 
 he wants to call forward to, if I have him forward to the extensions who then 
 forward to the real number, it works, thanks to adding Answer() to the 
 dialplan.
 
 -Gerard
 
 
 On 02/26/13 13:19, Gerard wrote:
 Hi everyone,

 I'm having a hard time figuring this issue out, we just switched from 
 a
 T1 PRI to a SIP trunk provider and that's when the issue started.
 Now when someone forwards all calls on their phone to a cellphone, 
 when a customer calls in, Asterisk correctly calls the cellphone and 
 connects the call, but there is a long delay before the audio starts, 
 basically for the first 6-10 seconds of the call there is dead 
 silence, eventually the audio will start and everything works correctly.
 We never had this problem with the PRI. So I suspect it has something 
 to do with a call coming in as SIP and going out as SIP.

 At first I thought it was a call forwarding issue because I got this 
 message in the console:
 [Feb 26 12:35:19] NOTICE[1143][C-025d]: app_dial.c:958 do_forward:
 Not accepting call completion offers from call-forward recipient
 Local/1XX@default-0013;1

 So I put this in my dial plan:

 1AA = {
 NoOp(${CALLERID(num)});
 Ringing;
 Set(CHANNEL(musicclass)=none);
 Dial(${OUTBOUND-TRUNKR}/1XX,30);
 Voicemail(198,u);
  };

 So basically as soon as someone calls incoming number AA, 
 Asterisk dials phone number XX. it's a quick and dirty way to 
 call forward.. and this does the same thing, there's a good 8 second 
 delay before the audio kicks in.


 There is a Linux firewall with NAT in the path, but I have no other 
 audio issues, so don't *think* it's a factor.
 I just upgraded to asterisk 11.2.1.


 Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
 2013-02-23 01:40:02 UTC


 Any help would be appreciated,
 Thanks,

 
 
 --
 _
 -- 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
 


-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

--
_
-- 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] Delay before audio starts

2013-03-01 Thread Leandro Dardini 
I think a simple tcpdump of the traffic will show the mystery. It can be
your provider doing something nasty. Have you tried using some other cheap
SIP termination? or arrange a fake termination yourself on another server?

Leandro

2013/3/1 Gerard gsara...@rarcoa.com

 I thought it was the re-invites too, but I have it turned off everywhere.

 On 03/01/13 08:36, Eric Wieling wrote:
  When Answer fixes the issue, the root cause is often NAT (could be
 firewall) since Answering the call prevents any reinvites.
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
  Sent: Friday, March 01, 2013 9:33 AM
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] Delay before audio starts
 
  I've found a workaround of sorts, If I change my below code to :
   1AA = {
   NoOp(${CALLERID(num)});
 Answer();  // --- add this
   Ringing;
   Set(CHANNEL(musicclass)=none);
   Dial(${OUTBOUND-TRUNKR}/1XX,30);
   Voicemail(198,u);
};
 
  That fixes the issue. It doesn't fix the call forward issue on the phone
 though. I've made a few extra extensions, one each corresponding to a
 number he wants to call forward to, if I have him forward to the extensions
 who then forward to the real number, it works, thanks to adding Answer()
 to the dialplan.
 
  -Gerard
 
 
  On 02/26/13 13:19, Gerard wrote:
  Hi everyone,
 
  I'm having a hard time figuring this issue out, we just switched from
  a
  T1 PRI to a SIP trunk provider and that's when the issue started.
  Now when someone forwards all calls on their phone to a cellphone,
  when a customer calls in, Asterisk correctly calls the cellphone and
  connects the call, but there is a long delay before the audio starts,
  basically for the first 6-10 seconds of the call there is dead
  silence, eventually the audio will start and everything works correctly.
  We never had this problem with the PRI. So I suspect it has something
  to do with a call coming in as SIP and going out as SIP.
 
  At first I thought it was a call forwarding issue because I got this
  message in the console:
  [Feb 26 12:35:19] NOTICE[1143][C-025d]: app_dial.c:958 do_forward:
  Not accepting call completion offers from call-forward recipient
  Local/1XX@default-0013;1
 
  So I put this in my dial plan:
 
  1AA = {
  NoOp(${CALLERID(num)});
  Ringing;
  Set(CHANNEL(musicclass)=none);
  Dial(${OUTBOUND-TRUNKR}/1XX,30);
  Voicemail(198,u);
   };
 
  So basically as soon as someone calls incoming number AA,
  Asterisk dials phone number XX. it's a quick and dirty way to
  call forward.. and this does the same thing, there's a good 8 second
  delay before the audio kicks in.
 
 
  There is a Linux firewall with NAT in the path, but I have no other
  audio issues, so don't *think* it's a factor.
  I just upgraded to asterisk 11.2.1.
 
 
  Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
  2013-02-23 01:40:02 UTC
 
 
  Any help would be appreciated,
  Thanks,
 
 
 
  --
  _
  -- 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
 


 --
 Gerard Saraber
 Network Admin.
 Rarcoa, Inc
 (630) 654-2580 x199
 (630) 654-3556 (fax)
 (630) 915-4122 (cell)

 --
 _
 -- 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

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_
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   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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