Re: [asterisk-users] No ringback heard

[Richard Mudgett]

On Thu, Aug 25, 2016 at 2:14 PM, Saint Michael  wrote:

> I dial two destination like this
>
> Dial(PJSIP/endpoint1/sip:${EXTEN}@${IPA}/endpoint1/sip:${EXTEN}@
> ${IPB})
>
> But I need the audio from one of them to be heard by the caller.
> None gets heard. I switch the order but nothing.
> How I get the audio for  one in particular?
>

You cannot.

In the general case, forked dials like that cannot pass any early media
back to the caller
because you would need to mix any early media from the two (or more)
outgoing
channels.  In addition, any mixed audio would be confusing.  Imaging
hearing "an all
circuits are busy" recording while at the same time hearing overlapping
ringback tones from
several other channels.

In specific cases, you are talking about a new feature which requires a
code change.

If what you want is ringback with a recording interspersed at intervals,
you can create a
music-on-hold class and have the caller hear that instead.

Richard
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Re: [asterisk-users] no ringback tone on outgoing call PRI line

[isrlgb]

https://issues.asterisk.org/view.php?id=18868

-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 8 May 2011 11:43:41 
To: asterisk-usersasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] no ringback tone on outgoing call PRI line

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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Steve Davies]

On 7 April 2011 23:04, Douglas Mortensen d...@impalanetworks.com wrote:
 Steve. Thanks for the insight. I won't pretend to know what early-audio is, 
 but I guess I'm about to find out :-).

 Also, I believe that I have a nearly identical setup like this with the exact 
 same SIP provider w/o any trouble. However, I think that system must be 
 running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to 
 confirm). Is there a significant difference between 1.2/1.4  1.6 in this 
 scenario?

 Thanks a million!! :-)

 -
 Doug Mortensen
 Network Consultant
 Impala Networks
 P: 505.327.7300
 .


 -Original Message-
 From: Steve Davies [mailto:davies...@gmail.com]
 Sent: Thursday, April 07, 2011 10:49 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] No ringback even though progressinband=yes is 
 set

 On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository:
 asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen


 In my personal experience with SIP and 1.6.x, that mostly depends on where 
 you are sending the call to. It depends on whether the next or subsequent leg 
 tries to use early-audio for the ring tone, or uses a Ringing event to signal 
 that is what is happening. It then depends on whether the originating 
 caller's equipment can understand early-audio ringing.

 We have a setup here where all our trunks support early-audio ringing except 
 one (an ISDN30 circuit) and we have to juggle things a bit sometimes to 
 ensure ringing occurs.

 Perhaps provide more details? Or you may find that tracing the SIP gives you 
 the clue that you need.

 Hope that helps,
 Steve



Early audio is audio that is sent before the call is answered,
usually in the form of a custom ring-tone or perhaps a cannot
connect, try later message. Some systems do not support it as it can
be abused to communicate at least basic information for free.

We had a problem with this when connecting Asterisk 1.2 to Asterisk
1.6 via IAX. A 1.2 SIP system will automatically switch into early
audio if it sees an early audio frame. 1.6 defaults to not doing this,
but there is a parameter to re-enable it. In this case we solved the
problem by upgrading to 1.6 everywhere :)

Regards,
Steve

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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Steve Davies]

On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository: 
 asterisk16-1.6.2.17.2-1_centos5
 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen


In my personal experience with SIP and 1.6.x, that mostly depends on
where you are sending the call to. It depends on whether the next or
subsequent leg tries to use early-audio for the ring tone, or uses a
Ringing event to signal that is what is happening. It then depends on
whether the originating caller's equipment can understand early-audio
ringing.

We have a setup here where all our trunks support early-audio ringing
except one (an ISDN30 circuit) and we have to juggle things a bit
sometimes to ensure ringing occurs.

Perhaps provide more details? Or you may find that tracing the SIP
gives you the clue that you need.

Hope that helps,
Steve

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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Sherwood McGowan]

On 4/7/2011 11:02 AM, Douglas Mortensen wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository: 
 asterisk16-1.6.2.17.2-1_centos5
 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen
 Network Consultant
 Impala Networks Inc
 CCNA, MCSA, Security+, A+
 Linux+, Network+, Server+
 .
 www.impalanetworks.com
 P: (505) 327-7300
 F: (505) 327-7545
 .


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If you are referring to a ringback tone when they first dial your
system, meaning that they immediately hear your IVR when they dial your
PBX's number, it's because that's how it's supposed to work. Unless you
tell your PBX to use the Ringing() app and wait for a period of time,
Asterisk normally picks up at the beginning of the IVR (since the first
thing you have to do to send audio via Background or Playback is issue
the command Answer() to start sending actual audio. (Note: The Ringing
app just signals RINGING to the remote party)

-- 
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant


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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Douglas Mortensen]

I have inbound calls going directly to a ring group. When callers call in, they 
(the callers) hear complete silence even though the phones that are part of the 
ring group ARE ringing properly. Employees can answer the calls when their 
phones ring, and everything works fine.

The problem is simply that the external caller never hears any ringing. Even if 
the SIP phones in the ring group ring for 5 rings, it is total silence even 
though there is ringing going on inside of the office.

I'm pretty sure it is a ringback issue.

I'm going to try to turn on SIP debugging  see what I can figure out that way. 
I do appreciate your help.

-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300
.

-Original Message-
From: Sherwood McGowan [mailto:sherwood.mcgo...@gmail.com] 
Sent: Thursday, April 07, 2011 12:36 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set

On 4/7/2011 11:02 AM, Douglas Mortensen wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository: 
 asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen
 Network Consultant
 Impala Networks Inc
 CCNA, MCSA, Security+, A+
 Linux+, Network+, Server+
 .
 www.impalanetworks.com
 P: (505) 327-7300
 F: (505) 327-7545
 .


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If you are referring to a ringback tone when they first dial your system, 
meaning that they immediately hear your IVR when they dial your PBX's number, 
it's because that's how it's supposed to work. Unless you tell your PBX to use 
the Ringing() app and wait for a period of time, Asterisk normally picks up at 
the beginning of the IVR (since the first thing you have to do to send audio 
via Background or Playback is issue the command Answer() to start sending 
actual audio. (Note: The Ringing app just signals RINGING to the remote party)

--
Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and 
PBX Solutions Consultant




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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Douglas Mortensen]

Steve. Thanks for the insight. I won't pretend to know what early-audio is, 
but I guess I'm about to find out :-).

Also, I believe that I have a nearly identical setup like this with the exact 
same SIP provider w/o any trouble. However, I think that system must be running 
asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is 
there a significant difference between 1.2/1.4  1.6 in this scenario?

Thanks a million!! :-)

-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300
.


-Original Message-
From: Steve Davies [mailto:davies...@gmail.com] 
Sent: Thursday, April 07, 2011 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set

On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository: 
 asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen


In my personal experience with SIP and 1.6.x, that mostly depends on where you 
are sending the call to. It depends on whether the next or subsequent leg tries 
to use early-audio for the ring tone, or uses a Ringing event to signal that is 
what is happening. It then depends on whether the originating caller's 
equipment can understand early-audio ringing.

We have a setup here where all our trunks support early-audio ringing except 
one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure 
ringing occurs.

Perhaps provide more details? Or you may find that tracing the SIP gives you 
the clue that you need.

Hope that helps,
Steve



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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Sherwood McGowan]

On 4/7/2011 4:54 PM, Douglas Mortensen wrote:
 I have inbound calls going directly to a ring group. When callers call in, 
 they (the callers) hear complete silence even though the phones that are part 
 of the ring group ARE ringing properly. Employees can answer the calls when 
 their phones ring, and everything works fine.

 The problem is simply that the external caller never hears any ringing. Even 
 if the SIP phones in the ring group ring for 5 rings, it is total silence 
 even though there is ringing going on inside of the office.

 I'm pretty sure it is a ringback issue.

 I'm going to try to turn on SIP debugging  see what I can figure out that 
 way. I do appreciate your help.

 -
 Doug Mortensen
 Network Consultant
 Impala Networks
 P: 505.327.7300

If you're using an interface (I believe you said AsteriskNOW), you might
want to check the Dial Options...Make sure that 'r' is one of the
options. The reason you're not hearing ringing is probably due to
Asterisk not sending a RINGING signal. If you have 'r' defined in the
dial options in your interface, then AsteriskNOW is probably using a
Dial command that is NOT using your global dial options.

-- 
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant


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Re: [asterisk-users] fake ringback tone

[wassim Darwish]

Hi:
Iam not using the 'r' option in my dial plan ,here what i have in my dial plan:
 
[gw]exten = _70.,1,Dial,SIP/grands/${EXTEN} Date: Fri, 9 Jan 2009 16:25:41 
-0500 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] fake ringback tone  On Fri, Jan 9, 2009 at 3:57 
PM, wassim Darwish wassim...@hotmail.com wrote:  hi:  When iam sending 
calls through sip a fake ringback tone is generated and  then call status 
can't be viewed (if call is ringing,busy,offline) it just  rings and rings. 
 Can i disable this?   Thanks in advance.   If you are using the r 
option in your Dial statement, remove it. That generates fake ringing. In 
FreePBX, that option is under the General settings, if plain jane Asterisk, 
just remove the r in your dial line.  --  Thanks, Steve Totaro 
+18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)  
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Re: [asterisk-users] fake ringback tone

[Philipp Kempgen]

wassim Darwish schrieb:
 Iam not using the 'r' option in my dial plan ,here what i have in my dial 
 plan:

Hint: Don't remove the line breaks:

 [gw]exten = _70.,1,Dial,SIP/grands/${EXTEN} Date: Fri, 9 Jan 2009 16:25:41 
 -0500 From: stot...@asteriskhelpdesk.com To: 
 asterisk-users@lists.digium.com Subject: Re: [asterisk-users] fake ringback 
 tone  On Fri, Jan 9, 2009 at 3:57 PM, wassim Darwish 
 wassim...@hotmail.com wrote:  hi:  When iam sending calls through sip a 
 fake ringback tone is generated and  then call status can't be viewed (if 
 call is ringing,busy,offline) it just  rings and rings.  Can i disable 
 this?   Thanks in advance.   If you are using the r option in your 
 Dial statement, remove it. That generates fake ringing. In FreePBX, that 
 option is under the General settings, if plain jane Asterisk, just remove 
 the r in your dial line.  --  Thanks, Steve Totaro +18887771888 (Toll 
 Free) +12409381212 (Cell) +12024369784 (Skype)  
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Re: [asterisk-users] fake ringback tone

[Steve Totaro]

On Fri, Jan 9, 2009 at 3:57 PM, wassim Darwish wassim...@hotmail.com wrote:
 hi:
 When iam sending calls through sip a fake ringback tone is generated and
 then call status can't be viewed (if call is ringing,busy,offline) it just
 rings and rings.
 Can i disable this?

 Thanks in advance.


If you are using the r option in your Dial statement, remove it.  That
generates fake ringing.  In FreePBX, that option is under the General
settings, if plain jane Asterisk, just remove the r in your dial line.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] Providing Ringback

[Grey Man]

Hi Igor,

We had an interconnect with a carrier that generated early media for
progress indications but the carrier's switch, in this case a Cerpack,
would only start sending the RTP for the early media AFTER it received
an RTP packet from the Asterisk end. Completely stupid behaviour since
early media is generally only one way but that's what it did.

We worked around it by recording 200ms of silence and playing that
back to the carrier's Cerpack with the Background command whenever we
received an incoming call. This got two way RTP set up and allowed the
progress tones to be correctly passed through to the user.

[noringback]
exten = _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n)
exten = _X.,2,Goto(incoming, ${EXTEN}, 1)

Regards,

Greyman.

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Re: [asterisk-users] Providing Ringback

[Igor Hernandez]

Thanks a lot Grey. I'll look into it.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com


Grey Man wrote:
 Hi Igor,
 
 We had an interconnect with a carrier that generated early media for
 progress indications but the carrier's switch, in this case a Cerpack,
 would only start sending the RTP for the early media AFTER it received
 an RTP packet from the Asterisk end. Completely stupid behaviour since
 early media is generally only one way but that's what it did.
 
 We worked around it by recording 200ms of silence and playing that
 back to the carrier's Cerpack with the Background command whenever we
 received an incoming call. This got two way RTP set up and allowed the
 progress tones to be correctly passed through to the user.
 
 [noringback]
 exten = _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n)
 exten = _X.,2,Goto(incoming, ${EXTEN}, 1)
 
 Regards,
 
 Greyman.
 
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Re: [asterisk-users] Bad ringback tone on zap channel

[Rob Hillis]

The ringback is coming from the Zap channel, since that's the 
destination of the call.  Therefore, the bad ring is more likely to be 
coming from the remote end.

What type of line are you making the call to?  Analogue?  E1/T1?  If 
it's analogue, I'd be guessing you have a faulty PSTN line.

James Lamanna wrote:
 Hmm ok.
 This was a call from a SIP phone registered with Asterisk outbound on
 a Zap trunk.
 So would Asterisk or the phone be generating the ringback tone in that case?

 It also happens very intermittently (maybe 1 in 10 calls at most...)

 -- James

 Rob Hillis wrote:
   
 In my experience, the ringback you get over a zap channel (be it
 analogue or digital) is generated by the remote end, /not/ Zaptel.

 The ringback you get over a SIP or IAX2 channel is often generated by
 either Asterisk or the SIP/IAX2 device you're calling from.


 James Lamanna wrote:
 
 Hi,
 I've noticed that sometimes instead of getting a regular ring tone
 when calling out on a Zap channel, I get this obnoxious loud noise
 which forces me to hang up.
 Is this a problem in the Zaptel driver? I seem to recall that ringback
 tones are generated by zaptel when dialing out from a SIP phone over a
 Zap trunk.
   

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Re: [asterisk-users] Bad ringback tone on zap channel

[Rob Hillis]

In my experience, the ringback you get over a zap channel (be it 
analogue or digital) is generated by the remote end, /not/ Zaptel.

The ringback you get over a SIP or IAX2 channel is often generated by 
either Asterisk or the SIP/IAX2 device you're calling from.


James Lamanna wrote:
 Hi,
 I've noticed that sometimes instead of getting a regular ring tone
 when calling out on a Zap channel, I get this obnoxious loud noise
 which forces me to hang up.
 Is this a problem in the Zaptel driver? I seem to recall that ringback
 tones are generated by zaptel when dialing out from a SIP phone over a
 Zap trunk.

 Thanks.

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 !DSPAM:484975fe67791857117240!


   

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Re: [asterisk-users] Bad ringback tone on zap channel

[James Lamanna]

Hmm ok.
This was a call from a SIP phone registered with Asterisk outbound on
a Zap trunk.
So would Asterisk or the phone be generating the ringback tone in that case?

It also happens very intermittently (maybe 1 in 10 calls at most...)

-- James

Rob Hillis wrote:
 In my experience, the ringback you get over a zap channel (be it
 analogue or digital) is generated by the remote end, /not/ Zaptel.

 The ringback you get over a SIP or IAX2 channel is often generated by
 either Asterisk or the SIP/IAX2 device you're calling from.


 James Lamanna wrote:
 Hi,
 I've noticed that sometimes instead of getting a regular ring tone
 when calling out on a Zap channel, I get this obnoxious loud noise
 which forces me to hang up.
 Is this a problem in the Zaptel driver? I seem to recall that ringback
 tones are generated by zaptel when dialing out from a SIP phone over a
 Zap trunk.

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Re: [asterisk-users] MixMonitor RingBack Tone Issue

[Giorgio Incantalupo]

Hi Jean-Marc,
I tried to use mixmonitor and seems that it works good. My problem is 
about calls after a transfer: it seems that asterisk can completely 
record a call in one file, only in case of blind transfer. 
If I make an attended transfer I have 2 or more sound files which are 
impossible to join.

Have you successfully recorded sound files of transfered calls in one file??

TIA

Giorgio Incantalupo


Jean-Marc Salsa wrote:

Indeed, perfect !
 
Thanks a lot ...
 
JM


 
On 2/17/07, *Trevor Peirce* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Jean-Marc Salsa wrote:

 exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r
 mailto: SIP/[EMAIL PROTECTED]
mailto:SIP/[EMAIL PROTECTED],30,r)

 Everything works perfectly, except when the softswitch, or the PSTN
 sends back RingBack Tone.

 I can see the RTP flow arriving to Asterisk,
 but, it seems that Asterisk doesn't forward it to the other party
 (next-hop).
Yes because you have the r in there, asterisk sends its own ringing.
If you want ringing to be heard from the PSTN, you need to leave that
option disabled.
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Re: [asterisk-users] MixMonitor RingBack Tone Issue

[Jean-Marc Salsa]

Indeed, perfect !

Thanks a lot ...

JM


On 2/17/07, Trevor Peirce [EMAIL PROTECTED] wrote:


Jean-Marc Salsa wrote:

 exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r
 mailto:SIP/[EMAIL PROTECTED],30,r)

 Everything works perfectly, except when the softswitch, or the PSTN
 sends back RingBack Tone.

 I can see the RTP flow arriving to Asterisk,
 but, it seems that Asterisk doesn't forward it to the other party
 (next-hop).
Yes because you have the r in there, asterisk sends its own ringing.
If you want ringing to be heard from the PSTN, you need to leave that
option disabled.
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Re: [asterisk-users] MixMonitor RingBack Tone Issue

[Trevor Peirce]

Jean-Marc Salsa wrote:


exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r 
mailto:SIP/[EMAIL PROTECTED],30,r)
 
Everything works perfectly, except when the softswitch, or the PSTN 
sends back RingBack Tone.
 
I can see the RTP flow arriving to Asterisk,
but, it seems that Asterisk doesn't forward it to the other party 
(next-hop).
Yes because you have the r in there, asterisk sends its own ringing.  
If you want ringing to be heard from the PSTN, you need to leave that 
option disabled.

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Re: [asterisk-users] Callback/ringback

[Lee Jenkins]

Yehavi Bourvine +972-8-9489444 wrote:

Enclosed bellow is the fragment from extenstions.conf which does two things:

*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.

   Regards, __Yehavi:



That's a very nice little script.

--

Warm Regards,

Lee

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RE: [asterisk-users] Callback/ringback

[Richard Soderblom]

Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206



Excellent little script. Thanks, Yehavi.



Best Regards

Richard Soderblom
Network Configurations
Cell: 
E-Mail: [EMAIL PROTECTED]



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-Original Message-

From: Yehavi Bourvine +972-8-9489444 [mailto:[EMAIL PROTECTED] 
Sent: 18 January 2007 07:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Callback/ringback

Enclosed bellow is the fragment from extenstions.conf which does two
things:

*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.

   Regards, __Yehavi:

; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension
dialled.
exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.

; Save the caller number at the called extension for *42 usage.
exten = _999XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _999XX,n,Set(DB(${From}/LastCalled)=${To})

; Now dial the extension.
exten = _999XX,n,Dial(SIP/${EXTEN},20,)   ; Dial the phone for 20
seconds.
; No answer or busy
exten = _999XX,n,GoTo(s-${DIALSTATUS},1)   ; Jump according to the
failure mode
exten = _999XX,n,Hangup()  ; Just to be sure...

; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call
to there

; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to
there

; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);


; Called here when the call is successfull and the user hanged the
phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for
us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp}  ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make
the call.
exten = h,6,System(echo Channel:  SIP/${tmp}  /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20  /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From}  /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\ 
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To}
/var/spool/asterisk/outgoing/)

exten = h,103,NoOp(Nothing to call)

; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)

; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
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Re: [asterisk-users] Callback/ringback

[Lee Jenkins]

Richard Soderblom wrote:

Hi.

Has anyone had any success in implementing a callback or ringback
function in Asterisk?

I've had a look at the callback-voicemail example on voip-info.org
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
However it won't quite work for me.

I need it for local SIP users which most of them don't have voicemail.
If one SIP user calls another SIP user and the second user is busy or
unavailable then Asterisk should inform the first user that the number
they dialed is busy and hangup the call.

Once the second caller is available again then Asterisk should initiate
a call back to both the users and connect them.

Any ideas on how to achieve this will be appreciated.


Richard,

That shouldn't be too difficult to do.  I recently wrote an agi binary 
that does nag calling for me which I think is related to what you want 
to do except that I am doing more calling out of the system.  Maybe 
deadagi could work?


Here is it's use in a AEL macro I'm working on:
http://www.datatrakpos.com/pos/datatalk/images/nagcall.htm

The AGI (nagcall) simply takes some parameters and uses them to create a 
.call file.  See:


http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

There is even an example on the page above that shows using the linux 
touch command to schedule the call to take place at a later time, 
although I have not successfully done this yet...still trying.


The biggest difference is that you will need a way to monitor the called 
extension to trigger a call back to the original caller using maybe 
deadagi or .call files?


http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+API

I'm pretty new to asterisk myself so there may be (probably are) other 
ways to do this, but this is where I would start poking around.


--

Warm Regards,

Lee

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Re: [asterisk-users] Callback/ringback

[Yehavi Bourvine +972-8-9489444]

Enclosed bellow is the fragment from extenstions.conf which does two things:

*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.

   Regards, __Yehavi:

; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled.
exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.

; Save the caller number at the called extension for *42 usage.
exten = _999XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _999XX,n,Set(DB(${From}/LastCalled)=${To})

; Now dial the extension.
exten = _999XX,n,Dial(SIP/${EXTEN},20,)   ; Dial the phone for 20 seconds.
; No answer or busy
exten = _999XX,n,GoTo(s-${DIALSTATUS},1)   ; Jump according to the failure 
mode
exten = _999XX,n,Hangup()  ; Just to be sure...

; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call to there

; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to there

; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);


; Called here when the call is successfull and the user hanged the phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp}  ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make the call.
exten = h,6,System(echo Channel:  SIP/${tmp}  /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20  /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From}  /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\  
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/)

exten = h,103,NoOp(Nothing to call)

; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)

; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()

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Re: [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP

[Tim Panton]

On 17 Apr 2006, at 00:30, Steve Feinstein wrote:

Actually it makes no difference.  I tried it in an attempt to get  
something to happen.


Thanks,
-Steve

Eric ManxPower Wieling wrote:
What happens if you remove the r option?  r is almost NEVER  
useful.


Steve Feinstein wrote:

I've been pulling my hair out over this one trying to understand it.

If you have a very simple extension:

exten = 1,n,Dial(IAX2/Steve|24|r)

Everything I've seen says this should tell the IAX phone (our own  
iaxclient based one) to make a ringing sound, or asterisk should  
make the ringback indication itself if it determines that the  
channel can't do it for itself.


But you can dial this extension all day and you never hear a  
ringback indication.  Dial it from a SIP softphone and you do.   
If you change the default country in the indications.conf, the  
SIP phone will change the way the ring sounds.  IAX, still nothing.


You can use PlayTones(ring) in the dialplan before the Dial(),  
and it seems to behave ok.  Playing the appropriate ring  
indication until the call is answered.  But it seems like the  
behavior is inconsistent with IAX vs. SIP.  Is this by design?


All the IAX soft phones I've tried are based on the same  
iaxclient libs, so it's hard to know if it's the phone or  
asterisk that's not behaving right.  Has anyone used an iax hard  
phone, some other IAX device/software, and does it exhibit the  
same behavior?  Or is this a problem with the iax code not being  
telling asterisk that IAX phones need to have their indications  
faked.


Any ideas about what's going on would be most gratefully  
appreciated.





I don't know the IAXclient libs, but an IAX client is supposed to  
send a RINGING packet back after it
accepts a call to notify the other end it should generate ringback  
for the user. The protocol allows it to go
straight to ANSWER, or send a PROCEEDING if it hasn't reached the end- 
point yet.


Is your client sending a RINGING packet at the right moment ?

Is there a call you should make (after accept but before answer) to  
get it to send RINGING?



Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP

[Eric \ManxPower\ Wieling]

What happens if you remove the r option?  r is almost NEVER useful.

Steve Feinstein wrote:

I've been pulling my hair out over this one trying to understand it.

If you have a very simple extension:

exten = 1,n,Dial(IAX2/Steve|24|r)

Everything I've seen says this should tell the IAX phone (our own 
iaxclient based one) to make a ringing sound, or asterisk should make 
the ringback indication itself if it determines that the channel can't 
do it for itself.


But you can dial this extension all day and you never hear a ringback 
indication.  Dial it from a SIP softphone and you do.  If you change the 
default country in the indications.conf, the SIP phone will change the 
way the ring sounds.  IAX, still nothing.


You can use PlayTones(ring) in the dialplan before the Dial(), and it 
seems to behave ok.  Playing the appropriate ring indication until the 
call is answered.  But it seems like the behavior is inconsistent with 
IAX vs. SIP.  Is this by design?


All the IAX soft phones I've tried are based on the same iaxclient libs, 
so it's hard to know if it's the phone or asterisk that's not behaving 
right.  Has anyone used an iax hard phone, some other IAX 
device/software, and does it exhibit the same behavior?  Or is this a 
problem with the iax code not being telling asterisk that IAX phones 
need to have their indications faked.


Any ideas about what's going on would be most gratefully appreciated.



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Re: [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP

[Steve Feinstein]

Actually it makes no difference.  I tried it in an attempt to get 
something to happen.


Thanks,
-Steve

Eric ManxPower Wieling wrote:

What happens if you remove the r option?  r is almost NEVER useful.

Steve Feinstein wrote:

I've been pulling my hair out over this one trying to understand it.

If you have a very simple extension:

exten = 1,n,Dial(IAX2/Steve|24|r)

Everything I've seen says this should tell the IAX phone (our own 
iaxclient based one) to make a ringing sound, or asterisk should make 
the ringback indication itself if it determines that the channel 
can't do it for itself.


But you can dial this extension all day and you never hear a ringback 
indication.  Dial it from a SIP softphone and you do.  If you change 
the default country in the indications.conf, the SIP phone will 
change the way the ring sounds.  IAX, still nothing.


You can use PlayTones(ring) in the dialplan before the Dial(), and it 
seems to behave ok.  Playing the appropriate ring indication until 
the call is answered.  But it seems like the behavior is inconsistent 
with IAX vs. SIP.  Is this by design?


All the IAX soft phones I've tried are based on the same iaxclient 
libs, so it's hard to know if it's the phone or asterisk that's not 
behaving right.  Has anyone used an iax hard phone, some other IAX 
device/software, and does it exhibit the same behavior?  Or is this a 
problem with the iax code not being telling asterisk that IAX phones 
need to have their indications faked.


Any ideas about what's going on would be most gratefully appreciated.



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RE: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections

[Brian C. Fertig]

are you giving answer()?

..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] No ringback tone generated by Asterisk with
OH323connections

I am using Asterisk (Debian unstable packages) with an OH323 connection
to my 
provider. Everything is working except for the generation of ringback
tones 
when I receive inbound calls from the PSTN. My provider tells me that
we're 
sending call progress indications and that because of this they're
expecting 
us to generate the ringback tone. Does anybody know how to configure
this in 
Asterisk? The relevant settings in oh323.conf are:

[general]
listenAddress=0.0.0.0
listenPort=1720
tcpStart=20001
tcpEnd=3
udpStart=20001
udpEnd=3
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
inBandDTMF=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
bandwidthLimit=2000
gatekeeper=DISABLE
gatekeeperTTL=600
userInputMode=RFC2833

The package versions I'm using are:

asterisk1.0.9.dfsg-5
asterisk-oh323  0.6.6pre3-4
libopenh323-1.15.3c21.15.3-4

-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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Re: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections

[Juan Jose Comellas]

No, I wasn't. I can't believe I made that stupid mistake. It started working 
after I added the call to Answer().

Thanks for your help. 


On Friday 30 September 2005 11:53, Brian C. Fertig wrote:
 are you giving answer()?

 ..o---o..
 Brian Fertig
 Network/Systems Engineer
 IT Administrator
 Planet Telecom, Inc.
 Tampa,FL Office

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
 Comellas
 Sent: Friday, September 30, 2005 10:32 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] No ringback tone generated by Asterisk with
 OH323connections

 I am using Asterisk (Debian unstable packages) with an OH323 connection
 to my
 provider. Everything is working except for the generation of ringback
 tones
 when I receive inbound calls from the PSTN. My provider tells me that
 we're
 sending call progress indications and that because of this they're
 expecting
 us to generate the ringback tone. Does anybody know how to configure
 this in
 Asterisk? The relevant settings in oh323.conf are:

 [general]
 listenAddress=0.0.0.0
 listenPort=1720
 tcpStart=20001
 tcpEnd=3
 udpStart=20001
 udpEnd=3
 fastStart=yes
 h245Tunnelling=yes
 h245inSetup=yes
 inBandDTMF=no
 jitterMin=20
 jitterMax=100
 ipTos=none
 outboundMax=10
 inboundMax=10
 simultaneousMax=10
 bandwidthLimit=2000
 gatekeeper=DISABLE
 gatekeeperTTL=600
 userInputMode=RFC2833

 The package versions I'm using are:

 asterisk  1.0.9.dfsg-5
 asterisk-oh3230.6.6pre3-4
 libopenh323-1.15.3c2  1.15.3-4

-- 
Juan Jose Comellas
([EMAIL PROTECTED])

-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Brian Dingman]

Found this info on their website:

http://www.livevoip.com/index.php?subject=2content=networkStatus

LiveVoip Operations Staff



DTMF - Ringback Issues

Currently, Asterisk is using the timing of the input stream to
reproduce the output stream. This means that when no RTP streams are
being sent from the peer Endpoint Gateway, Asterisk is unable to
generate audio. This approach or limitation leads to one way speech
conditions. Plus - Some devices don't generate audio until the answer
supervision is received from the called. For all these scenarios, no
ringback can be presented to the calling party. In cases where the
endpoints are using silence compression, the audio from asterisk is
chopped. Its fine if your run Asterisk with a T-1 Card, if not then
you are going to experience issues.

What Can or Should be Done?

To get this solved, Asterisk should obtain its clocking from an
internal source in a way that an output stream can be generated
without getting any RTP input. The clocking should then be taken from
an internal timing mechanism that keeps track of the synchronization.
The solution should not require T1 connectivity [IE: no TDM hardware].
Such T1 connectivity would severely limit traffic on the LiveVoip
Global SIP network via IP. Developers should work to solve the no
alerting scenario's [when peer is set in RCV only mode] and all issues
related to the use of silence compression. A configuration option
should exist to choose the timing method for customers that want to
use Asterisk in calling card applications or any application where no
T-1 cards will ever be required.

Status:

LiveVoip engineers have developed a workaround for our internal switch
network. This will be tested and could take up to 14 days to install
in every LiveVoip Network Node location.


On Tue, 15 Mar 2005 17:07:53 -0500, Robert Webb [EMAIL PROTECTED] wrote:
 
 On Tue, 15 Mar 2005 14:50:38 -0700
   Daniel Webb [EMAIL PROTECTED] wrote:
  On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk
 wrote:
 
  Dude, where have you been?  This has been discussed here
 at length.
  Everyone agrees that it's on LiveVOIP's end, but they're
 shrugging their
  shoulders and pointing toward *.  Search the list.
 
  Could you point out the best way to search the list?
 
  Perhaps go to
 http://lists.digium.com/pipermail/asterisk-users/, go to
  each month one at a time, then click threads, then do
 a page search?
  What a swell interface.
 
 How about learning a few Google skills and in the search
 line type:
 
 site:lists.digium.com search criteria
 
 THe above site command will only search the url specified.
 In this case the Asterisk lists.
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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Daniel Webb]

On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote:

 Dude, where have you been?  This has been discussed here at length.
 Everyone agrees that it's on LiveVOIP's end, but they're shrugging their
 shoulders and pointing toward *.  Search the list.

Could you point out the best way to search the list?

Perhaps go to http://lists.digium.com/pipermail/asterisk-users/, go to
each month one at a time, then click threads, then do a page search?
What a swell interface.
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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Mohit Muthanna]

Try googling:

QUERY: Asterisk-Users Search String

Works quite well.

--
www.justfuckinggoogleit.com


On Tue, 15 Mar 2005 14:50:38 -0700, Daniel Webb [EMAIL PROTECTED] wrote:
 On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote:
 
  Dude, where have you been?  This has been discussed here at length.
  Everyone agrees that it's on LiveVOIP's end, but they're shrugging their
  shoulders and pointing toward *.  Search the list.
 
 Could you point out the best way to search the list?
 
 Perhaps go to http://lists.digium.com/pipermail/asterisk-users/, go to
 each month one at a time, then click threads, then do a page search?
 What a swell interface.
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who don't.
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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Robert Webb]

On Tue, 15 Mar 2005 14:50:38 -0700
 Daniel Webb [EMAIL PROTECTED] wrote:
On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk 
wrote:

Dude, where have you been?  This has been discussed here 
at length.
Everyone agrees that it's on LiveVOIP's end, but they're 
shrugging their
shoulders and pointing toward *.  Search the list.
Could you point out the best way to search the list?
Perhaps go to 
http://lists.digium.com/pipermail/asterisk-users/, go to
each month one at a time, then click threads, then do 
a page search?
What a swell interface.

How about learning a few Google skills and in the search 
line type:

site:lists.digium.com search criteria
THe above site command will only search the url specified. 
In this case the Asterisk lists.
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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Rich Adamson]

  If asterisk is going to be modified to support LiveVoip expectations,
  then yet another Dial option would need to be implemented to
  force ringback to occur as an audio stream for iax only. Guess
  one could open a bug report for both LiveVoip and Asterisk, but
  not likely to be addressed any time soon since this is itsp
  dependent.
 
 This would actually be a good idea to to do, simply provide a switch 
 different than 'r' that forces an audio string ringback.  Not only would it 
 work for LiveVoip, but for other circumstances as well.
 
 Sure, one could argue that we're creating a band-aid for a problem that 
 shouldn't have to exist, but hey, if it works then I don't see why not add 
 it in.  People don't need to use the feature if they don't want to... :)

FWIW, I submitted a detailed description of the ringback problem to
[EMAIL PROTECTED] this morning (March 13).

After personally analyzing the ringback issue (and without any feedback
from livevoip as yet), my best estimate is that livevoip is going to
consider this an asterisk issue as the call is considered answered
(due to asterisk's IVR answering the call), and that any further
ringing feedback should come from the end nodes (not from the middle-
man livevoip switch). That is basically how all real telephony (non-
voip providers) would handle such things.

That would imply that we'll need to submit a feature request for
asterisk to support some sort of inband ringback option selectable
by the asterisk implementor.

Rich


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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Rich Adamson]

 Grett.  This should be loads of fun then...  8(
 
 I have noticed what I can only describe a negative undertone with
 several VoIP poviders.
 Not an easy customer?  We don't want you.  Things like that.
 The LiveVoIP website is in fact like that.  
 
 There are several places on the site that just flat out say are you
 customer type x?, we don't want you then.
 Not my way of doing business but to each their own.  I guess as long as
 they service my account and provide a good voip connection it won't mean
 much to me.

There are more then a few folks using * that try various itsp services
without a clue as to how to make things work, and/or with unrealistic
expectations. I happen to like their no-nonsense approach on their
web site of getting your attention. It sort of resembles some of the
postings on this list relative to 'did you try to look for doc'.

Overall, I'd give livevoip high marks in both quality and service. Let's
hope they can keep it up. :)


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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Rich Adamson]

  I saw some coverage of this in the list archive but no 
 one seems to have
  posted a resolution.
  
  I am using [EMAIL PROTECTED] 0.06 and when I get a call from 
 LiveVoip over
  IAX I dump it into my IVR.
  From there the call is routed to groups based upon 
 input.
  
  However, there is no ringback indicated to the IAX 
 caller.
  
  Perhaps they expect you to provide audioable progress 
 information inband 
  on the reverse channel? I.e. use the 'r' option on the 
 dial command etc. 
  That is the way some isdn lines etc work.
  
  Peter
  
 
 That is what everyone is bitching about. No matter whether 
 you use the r option or not, you never get ringback 
 through LiveVoIP. And they consistently point the finger 
 at * rather that trying to solve the problem.

If you use ethereal to inspect the iax packets in the above
case, you see that asterisk is sending an IAX Type=Control packet
with a Control subclass: Ringing (3) to the LiveVoip switch.

LiveVoip is ignoring that particular control packet. I'd have
to guess that LiveVoip wants ringback to occur in the audio
stream, not as a iax control packet, and therefore is blaming 
asterisk.

The r option within asterisk (in the above case) is doing
exactly what Mark intended for asterisk-to-asterisk iax
connections, which is different then LiveVoip expectations.
So, who is wrong here, or is this just human translations of
what is expected in a non-rfc communications environment?

If asterisk is going to be modified to support LiveVoip expectations,
then yet another Dial option would need to be implemented to
force ringback to occur as an audio stream for iax only. Guess
one could open a bug report for both LiveVoip and Asterisk, but 
not likely to be addressed any time soon since this is itsp 
dependent.

Rich


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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Wiley Siler]

Title: RE: [Asterisk-Users] No ringback over IAX - LiveVoip






Excellent thing to 
hear. I am glad there are positives on this site as well as teh 
warnings.

Now to get the ringback issue 
resolved

Using m switch to get MOH is OK but there 
has to be alogical reason this is occuring adn a way to resolve.

Thanks,
Wiley



From: [EMAIL PROTECTED] on 
behalf of Rich AdamsonSent: Sat 3/12/2005 5:49 AMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[Asterisk-Users] No ringback over IAX - LiveVoip

 Grett. This should be loads of fun then... 
8( I have noticed what I can only describe a negative undertone 
with several VoIP poviders. Not an easy customer? We don't 
want you. Things like that. The LiveVoIP website is in fact like 
that. There are several places on the site that just flat 
out say "are you customer type x?, we don't want you then". Not 
my way of doing business but to each their own. I guess as long as 
they service my account and provide a good voip connection it won't mean 
much to me.There are more then a few folks using * that try various itsp 
serviceswithout a clue as to how to make things work, and/or with 
unrealisticexpectations. I happen to like their "no-nonsense" approach on 
theirweb site of getting your attention. It sort of resembles some of 
thepostings on this list relative to 'did you try to look for 
doc'.Overall, I'd give livevoip high marks in both quality and service. 
Let'shope they can keep it up. 
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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Robert Webb]

On Fri, 11 Mar 2005 11:47:53 -0700
 Wiley Siler [EMAIL PROTECTED] wrote:
Hello All,
I saw some coverage of this in the list archive but no 
one seems to have
posted a resolution.

I am using [EMAIL PROTECTED] 0.06 and when I get a call from 
LiveVoip over
IAX I dump it into my IVR.
From there the call is routed to groups based upon input.
However, there is no ringback indicated to the IAX 
caller.

Does anyone know how to resolve this problem?
Thanks,
Wiley
According to LiveVoip folks it is ASterisk's fault that 
this does not work. Even though there are many of us using 
IAX with other providers with no problems.

Just don't complain too much as they will email you and 
tell you that if you do not like it to just cancel. Or, as 
one user noted here, it seemed that his account was just 
deleted without any contact. Not sure if that is really 
what happened or not. But the email contact telling you to 
cancel will. They did it to me..

Robert
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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Wiley Siler]

Grett.  This should be loads of fun then...  8(

I have noticed what I can only describe a negative undertone with
several VoIP poviders.
Not an easy customer?  We don't want you.  Things like that.
The LiveVoIP website is in fact like that.  

There are several places on the site that just flat out say are you
customer type x?, we don't want you then.
Not my way of doing business but to each their own.  I guess as long as
they service my account and provide a good voip connection it won't mean
much to me.

Thanks,
Wiley




-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 11, 2005 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Wiley Siler
Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip


On Fri, 11 Mar 2005 11:47:53 -0700
  Wiley Siler [EMAIL PROTECTED] wrote:
 Hello All,
 
 I saw some coverage of this in the list archive but no one seems to 
have  posted a resolution.
 
 I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over

IAX I dump it into my IVR.
From there the call is routed to groups based upon input.
 
 However, there is no ringback indicated to the IAX caller.
 
 Does anyone know how to resolve this problem?
 
 Thanks,
 Wiley
 

According to LiveVoip folks it is ASterisk's fault that this does not
work. Even though there are many of us using IAX with other providers
with no problems.

Just don't complain too much as they will email you and tell you that if
you do not like it to just cancel. Or, as one user noted here, it seemed
that his account was just deleted without any contact. Not sure if that
is really what happened or not. But the email contact telling you to
cancel will. They did it to me..

Robert
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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Peter Svensson]

On Fri, 11 Mar 2005, Wiley Siler wrote:

 I saw some coverage of this in the list archive but no one seems to have
 posted a resolution.
 
 I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over
 IAX I dump it into my IVR.
 From there the call is routed to groups based upon input.
 
 However, there is no ringback indicated to the IAX caller.

Perhaps they expect you to provide audioable progress information inband 
on the reverse channel? I.e. use the 'r' option on the dial command etc. 
That is the way some isdn lines etc work.

Peter


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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Wiley Siler]

I am pretty sure that AAH is adding that automatically.  I am checking
it out right now.

Thank you for the poitner!

Chers,
Wiley 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Friday, March 11, 2005 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip

On Fri, 11 Mar 2005, Wiley Siler wrote:

 I saw some coverage of this in the list archive but no one seems to 
 have posted a resolution.
 
 I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over

 IAX I dump it into my IVR.
 From there the call is routed to groups based upon input.
 
 However, there is no ringback indicated to the IAX caller.

Perhaps they expect you to provide audioable progress information inband
on the reverse channel? I.e. use the 'r' option on the dial command etc.

That is the way some isdn lines etc work.

Peter


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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Robert Webb]

On Fri, 11 Mar 2005 21:48:07 +0100 (CET)
 Peter Svensson [EMAIL PROTECTED] wrote:
On Fri, 11 Mar 2005, Wiley Siler wrote:
I saw some coverage of this in the list archive but no 
one seems to have
posted a resolution.

I am using [EMAIL PROTECTED] 0.06 and when I get a call from 
LiveVoip over
IAX I dump it into my IVR.
From there the call is routed to groups based upon 
input.

However, there is no ringback indicated to the IAX 
caller.
Perhaps they expect you to provide audioable progress 
information inband 
on the reverse channel? I.e. use the 'r' option on the 
dial command etc. 
That is the way some isdn lines etc work.

Peter
That is what everyone is bitching about. No matter whether 
you use the r option or not, you never get ringback 
through LiveVoIP. And they consistently point the finger 
at * rather that trying to solve the problem.
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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Wiley Siler]

Hmm... It could be a simple mistake of course.

Is it safe to assume that the host server on their side is Asterisk as
well?
I assume there are no telco level devices that offer IAX.  Could be
wrong.

If that is the case, it should be easy to infer what is wrong on their
side.
Obviously someone here has mated a couple of Asterisk boxes before.
What would have to be set wrong on their side for the ring tone not to
pass?

Once that is known, LiveVoIP users can send them the request to fix that
particular problem.

Thanks!
Wiley


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Friday, March 11, 2005 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip


On Fri, 11 Mar 2005 21:48:07 +0100 (CET)
  Peter Svensson [EMAIL PROTECTED] wrote:
 On Fri, 11 Mar 2005, Wiley Siler wrote:
 
 I saw some coverage of this in the list archive but no one seems to 
have  posted a resolution.
 
 I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip 
over  IAX I dump it into my IVR.
 From there the call is routed to groups based upon
input.
 
 However, there is no ringback indicated to the IAX caller.
 
 Perhaps they expect you to provide audioable progress information 
inband  on the reverse channel? I.e. use the 'r' option on the dial 
command etc.
 That is the way some isdn lines etc work.
 
 Peter
 

That is what everyone is bitching about. No matter whether you use the
r option or not, you never get ringback through LiveVoIP. And they
consistently point the finger at * rather that trying to solve the
problem.
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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Jay Milk]

Maybe the way they present this selection isn't very fortunate, but I
fully understand where they're coming from.  They're providing wholesale
VOIP at wholesale prices.

Seriously, do the math -- If you're buying an inbound 800# from them,
you could get away with spending only $2.50/month ($29.95 prepaid,
expires in a year).  And if you used these $2.50 to receive 200 incoming
minutes a month, they'll make maybe $1 on you (assuming 1.27c/min sale
vs. 0.8c/min cost).

If it takes them more than ONE MINUTE/MONTH to support you, they've
already blown their profit.

I myself have often walked away from expensive customers, and business
people much smarter than me do this on a daily basis.

 -Original Message-
 From: Wiley Siler [mailto:[EMAIL PROTECTED] 
 Sent: Friday, March 11, 2005 1:53 PM
 To: Robert Webb; Asterisk Users Mailing List - Non-Commercial 
 Discussion
 Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
 
 
 Grett.  This should be loads of fun then...  8(
 
 I have noticed what I can only describe a negative undertone 
 with several VoIP poviders. Not an easy customer?  We don't 
 want you.  Things like that. The LiveVoIP website is in fact 
 like that.  
 
 There are several places on the site that just flat out say 
 are you customer type x?, we don't want you then. Not my 
 way of doing business but to each their own.  I guess as long 
 as they service my account and provide a good voip connection 
 it won't mean much to me.
 
 Thanks,
 Wiley

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Re: [Asterisk-Users] No ringback over IAX - LiveVoip

[Mike Benoit]

Using the 'r' option doesn't seem to make a difference for me. To work
around this issue, I'm using the 'm' option to play MOH. 

However it just occurred to me, does anyone have a .mp3 recording of a
phone ringing for 30+ seconds? I could play that instead of regular
music and it would probably work not too bad.


On Fri, 2005-03-11 at 21:48 +0100, Peter Svensson wrote:
 On Fri, 11 Mar 2005, Wiley Siler wrote:
 
  I saw some coverage of this in the list archive but no one seems to have
  posted a resolution.
  
  I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over
  IAX I dump it into my IVR.
  From there the call is routed to groups based upon input.
  
  However, there is no ringback indicated to the IAX caller.
 
 Perhaps they expect you to provide audioable progress information inband 
 on the reverse channel? I.e. use the 'r' option on the dial command etc. 
 That is the way some isdn lines etc work.
 
 Peter
 
 
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Mike Benoit [EMAIL PROTECTED]


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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Wiley Siler]

I agree totally.  I just have never seen anyone post it on their site so
brazenly.

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Friday, March 11, 2005 3:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip

Maybe the way they present this selection isn't very fortunate, but I
fully understand where they're coming from.  They're providing wholesale
VOIP at wholesale prices.

Seriously, do the math -- If you're buying an inbound 800# from them,
you could get away with spending only $2.50/month ($29.95 prepaid,
expires in a year).  And if you used these $2.50 to receive 200 incoming
minutes a month, they'll make maybe $1 on you (assuming 1.27c/min sale
vs. 0.8c/min cost).

If it takes them more than ONE MINUTE/MONTH to support you, they've
already blown their profit.

I myself have often walked away from expensive customers, and business
people much smarter than me do this on a daily basis.

 -Original Message-
 From: Wiley Siler [mailto:[EMAIL PROTECTED]
 Sent: Friday, March 11, 2005 1:53 PM
 To: Robert Webb; Asterisk Users Mailing List - Non-Commercial 
 Discussion
 Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
 
 
 Grett.  This should be loads of fun then...  8(
 
 I have noticed what I can only describe a negative undertone with 
 several VoIP poviders. Not an easy customer?  We don't want you.  
 Things like that. The LiveVoIP website is in fact like that.
 
 There are several places on the site that just flat out say are you 
 customer type x?, we don't want you then. Not my way of doing 
 business but to each their own.  I guess as long as they service my 
 account and provide a good voip connection it won't mean much to me.
 
 Thanks,
 Wiley

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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Jay Milk]

Dude, where have you been?  This has been discussed here at length.
Everyone agrees that it's on LiveVOIP's end, but they're shrugging their
shoulders and pointing toward *.  Search the list.

 -Original Message-
 From: Wiley Siler [mailto:[EMAIL PROTECTED] 
 Sent: Friday, March 11, 2005 3:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion; 
 [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
 
 
 Hmm... It could be a simple mistake of course.
 
 Is it safe to assume that the host server on their side is 
 Asterisk as well? I assume there are no telco level devices 
 that offer IAX.  Could be wrong.
 
 If that is the case, it should be easy to infer what is wrong 
 on their side. Obviously someone here has mated a couple of 
 Asterisk boxes before. What would have to be set wrong on 
 their side for the ring tone not to pass?
 
 Once that is known, LiveVoIP users can send them the request 
 to fix that particular problem.
 
 Thanks!
 Wiley

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RE: [Asterisk-Users] No ringback over IAX - LiveVoip

[Wiley Siler]

I have been on the list and the archive as I stated.
Saw the emails but never saw any resolution given as I stated.

In case you did not notice, I am suggesting that we can figure out what
the problem on their side is and give them the resolution.
In that way the problem gets solved and we all benefit.  They may
honestly believe nothing is wrong. We should focus on solving the
problem.

Read the other posts before you issue a flame over checking the list.
Escpecially since you did not read the list yourself to see my other
post or even the fact that I am suggesting we find the problem.

I am absolutely fed up with people like you bouncing out these shitty
replies like you own the list.

If you don't like my question.  Delete it.
If you don't want to answer.  Don't answer.
If all you have to say is negative prattling, keep it to yourself.

Regards,
Wiley


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Friday, March 11, 2005 3:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip

Dude, where have you been?  This has been discussed here at length.
Everyone agrees that it's on LiveVOIP's end, but they're shrugging their
shoulders and pointing toward *.  Search the list.

 -Original Message-
 From: Wiley Siler [mailto:[EMAIL PROTECTED]
 Sent: Friday, March 11, 2005 3:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion; 
 [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
 
 
 Hmm... It could be a simple mistake of course.
 
 Is it safe to assume that the host server on their side is Asterisk as

 well? I assume there are no telco level devices that offer IAX.  Could

 be wrong.
 
 If that is the case, it should be easy to infer what is wrong on their

 side. Obviously someone here has mated a couple of Asterisk boxes 
 before. What would have to be set wrong on their side for the ring 
 tone not to pass?
 
 Once that is known, LiveVoIP users can send them the request to fix 
 that particular problem.
 
 Thanks!
 Wiley

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RE: [Asterisk-Users] No ringback over IAX - LiveVoip (or how Wiley made my /dev/null list)

[Jay Milk]

That's very nice and all, but not constructive.  Please DO read the
posts regarding LiveVOIP's ringback problem.  LiveVOIP has even
identified as * sending back a non-standard signal.  So, we know what
the problem is, we know it doesn't exist with other providers, and even
in the face of all this information, LiveVOIP seems to be refusing or
unable to fix it.

That said, there was nothing negative about my post, just stating the
very obvious fact that you may have missed part of the discussion.  I
didn't suggest why you may have missed it, nor did I imply that there's
anything wrong with you for missing it.  I do, however, take note that
you're taking this list much too seriously.  If you defend it so
vigorously, maybe you could take your arguments off-list rather than
plugging up communications with no less than 30 messages in the last 24
hours, the majority of which unconstructive bickering about your
personal disagreement with some jerk's response to a newbie.

I'm not defending the decision of some to be hostile towards those who
are lazy, or even the ridiculously thoughtless sentiment to complain
to 10,000 people about some wasted bytes -- but your posts and their
frequency are adding a lot of noise to a list which could otherwise be
useful.  Since you like to give out your recommendations as
bullet-points, let me add mine:

- If you don't know the answer to a question, don't answer it.
- If you have no new information to add, don't post.
- If you disagree with someone's attitude or post or whatever, tell that
ONE person; don't bother the rest of the free world with your scrupels.
- And lastly, don't get offended on this list -- it ain't worth it.
Spend more time with your family or pets or car or whatever makes you
smile.

Have a good weekend.

(and yes, you're dev/nul'ed -- it's nothing personal, I'm just trying to
cut down on the noise, and at your current rate, you'd add another 1,000
messages/month to this list)

 -Original Message-
 From: Wiley Siler [mailto:[EMAIL PROTECTED] 
 Sent: Friday, March 11, 2005 5:05 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
 
 
 I have been on the list and the archive as I stated.
 Saw the emails but never saw any resolution given as I stated.
 
 In case you did not notice, I am suggesting that we can 
 figure out what the problem on their side is and give them 
 the resolution. In that way the problem gets solved and we 
 all benefit.  They may honestly believe nothing is wrong. We 
 should focus on solving the problem.
 
 Read the other posts before you issue a flame over checking 
 the list. Escpecially since you did not read the list 
 yourself to see my other post or even the fact that I am 
 suggesting we find the problem.
 
 I am absolutely fed up with people like you bouncing out 
 these shitty replies like you own the list.
 
 If you don't like my question.  Delete it.
 If you don't want to answer.  Don't answer.
 If all you have to say is negative prattling, keep it to yourself.
 
 Regards,
 Wiley

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RE: [Asterisk-Users] No ringback

[Gene Kochanowsky]

Title: No ringback 








I had a similar problem. What I did what
checked out the version before 03-02-2004. Some change after that date is
causing the problem.



Gene











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie
Sent: Saturday, April 03, 2004
4:32 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] No
ringback







I
just configured Asterisk on a new machine, and other things seem to be working
fine, I don't get any audible ringback when dialing calls from a SIP phone or a
standard phone connected through a TDM400P. What am I doing wrong here?

Thanks


-brian

RE: [Asterisk-Users] No ringback

[Brian Cuthie]

Title: No ringback



Thanks. Actually,I got the latest from the cvs 
repository and it's fixed there, too. I suspect that it got broken at some point 
briefly before someone fixed it.

-brian

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Gene 
  KochanowskySent: Saturday, April 03, 2004 5:25 PMTo: 
  [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] No 
  ringback
  
  
  I had a similar 
  problem. What I did what checked out the version before 03-02-2004. Some 
  change after that date is causing the problem.
  
  Gene
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Brian CuthieSent: Saturday, April 03, 2004 4:32 
  PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] No 
  ringback
  
  
  I 
  just configured Asterisk on a new machine, and other things seem to be working 
  fine, I don't get any audible ringback when dialing calls from a SIP phone or 
  a standard phone connected through a TDM400P. What am I doing wrong 
  here?
  Thanks 
  
  -brian 
  

Re: [Asterisk-Users] Missing ringback tone on C7960

[Walker Haddock]

On Mon, Mar 22, 2004 at 01:48:05PM -0600, Rich Adamson wrote:
 Just upgraded to stable CVS-03/20/04-11:54:56 
 
 C7960 - * C7960 (all on the same wire), call from on phone
 to the other does not receive any ringback signal. Total silence
 while the phone is actually ringing, then hear voicemail anouncements
 after the 15 second timeout.
 
 Was working fine before the upgrade.
 
 Anyone else observe the same issue?
 
 Rich
I upgraded two servers to the v1-0_stable yesterday.  Both servers would not provide 
ring tone to the callers when I was dialing the 7960 phones.  I added the `r` option 
to the Dial command and that took care of it.

Walker
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
***
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Re: [Asterisk-Users] Missing ringback tone on C7960

[Rich Adamson]

 On Mon, Mar 22, 2004 at 01:48:05PM -0600, Rich Adamson wrote:
  Just upgraded to stable CVS-03/20/04-11:54:56 
  
  C7960 - * C7960 (all on the same wire), call from on phone
  to the other does not receive any ringback signal. Total silence
  while the phone is actually ringing, then hear voicemail anouncements
  after the 15 second timeout.
  
  Was working fine before the upgrade.
  
  Anyone else observe the same issue?
  
  Rich
 I upgraded two servers to the v1-0_stable yesterday.  Both servers would not provide 
 ring tone 
to the callers when I was dialing the 7960 phones.  I added the `r` option to the Dial 
command 
and that took care of it.
 

Thanks Walker... I had to do the same thing. Apparently something changed
in the sip area.

Rich


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RE: [Asterisk-Users] No Ringback on Iconnect

[Andrew Joakimsen]

What is the 

Exten = .Dial( 





Line from your extensions.conf?







-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
7:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect



When I place a call using Iconnecthere or Nikotel as my sip
provider, I hear no call progress such as ringback when making a call. If
I program the SIP phone to directly access iconnect or nikotel I do hear
ringing when the outbound call is placed. Does anyone
else have this problem or offer any suggestions? Thanks, Kevin

RE: [Asterisk-Users] No Ringback on Iconnect

[Kevin]

I have tried both of
these:



exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]

exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]||r





-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED] 
Sent: Sunday, October 05, 2003
7:56 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



What is the 

Exten =
.Dial( 





Line from your
extensions.conf?







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
7:23 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect



When I place a call using
Iconnecthere or Nikotel as my sip provider, I hear no call progress such as
ringback when making a call. If I program the SIP phone to directly
access iconnect or nikotel I do hear ringing when the outbound call is
placed. Does anyone else have this problem or offer any
suggestions? Thanks, Kevin

RE: [Asterisk-Users] No Ringback on Iconnect

[Andrew Joakimsen]

Try



exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)
or



exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)





It should work







-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
9:04 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



I have tried both of these:



exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]

exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]||r





-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED] 
Sent: Sunday, October 05, 2003
7:56 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



What is the 

Exten = .Dial(






Line from your
extensions.conf?







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
7:23 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect



When I place a call using
Iconnecthere or Nikotel as my sip provider, I hear no call progress such as
ringback when making a call. If I program the SIP phone to directly
access iconnect or nikotel I do hear ringing when the outbound call is
placed. Does anyone else have this problem or offer any
suggestions? Thanks, Kevin

RE: [Asterisk-Users] No Ringback on Iconnect

[Kevin]

Changed my conf
file to:



exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED],90,r



still no ringback



-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED] 
Sent: Sunday, October 05, 2003
9:20 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



Try



exten =
_1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r) or



exten =
_1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)





It should work







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
9:04 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



I have tried both of
these:



exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]

exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]||r





-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED] 
Sent: Sunday, October 05, 2003
7:56 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



What is
the 

Exten
= .Dial( 





Line
from your extensions.conf?







-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
7:23 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect



When I place a call using Iconnecthere
or Nikotel as my sip provider, I hear no call progress such as ringback when
making a call. If I program the SIP phone to directly access iconnect or
nikotel I do hear ringing when the outbound call is placed. Does anyone
else have this problem or offer any suggestions? Thanks, Kevin

RE: [Asterisk-Users] No Ringback on Iconnect

[Andrew Joakimsen]

Then remove the ,r and see if it works.
Have you setup your indications.conf? Do you get any messages from asterisk
CLI? What device are you using?







-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October
 05, 2003 11:01 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



Changed my conf file to:



exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED],90,r



still no ringback



-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED] 
Sent: Sunday, October
 05, 2003 9:20 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



Try



exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)
or



exten =
_1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)





It should work







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Sunday, October
 05, 2003 9:04 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



I have tried both of
these:



exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]

exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED]||r





-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED] 
Sent: Sunday, October
 05, 2003 7:56 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on Iconnect



What is
the 

Exten
= .Dial( 





Line
from your extensions.conf?







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Sunday, October
 05, 2003 7:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect



When I place a call using
Iconnecthere or Nikotel as my sip provider, I hear no call progress such as
ringback when making a call. If I program the SIP phone to directly
access iconnect or nikotel I do hear ringing when the outbound call is
placed. Does anyone else have this problem or offer any
suggestions? Thanks, Kevin

Re: [Asterisk-Users] HELP!!!! Ringback oh323

[Michael Ulitskiy]

Specify option 'r' to dial application.

Michael

On Friday 01 August 2003 07:13 pm, [EMAIL PROTECTED] wrote:
 Hi
 
  What command i need to use to make a call with oh323 and hear the
 ringback sound
 
 Thanks
 
 
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Re: [Asterisk-Users] false ringback

[Petr Michálek]

Michael Bielicki wrote:

Is it possible to give a false ringbakc on asterisk ?
 

What hardware are you using?

Petr Michalek

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Re: [Asterisk-Users] false ringback

[Michael Bielicki]

digium cards on * but the carrier connections are all h323 and some of them 
don't provide ringback.

On Thursday 03 Apr 2003 21:42, Petr Michálek shaped the electrons to say:
 Michael Bielicki wrote:
 Is it possible to give a false ringbakc on asterisk ?

 What hardware are you using?

 Petr Michalek

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-- 
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Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/

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