asterisk users - Jan 2005 - No ringback on IAX channel after selecting menu option

Here is the call flow:
[ivr-incoming]
exten => s,1,LookupCIDName
exten => s,2,DigitTimeout(2)
exten => s,3,ResponseTimeout(10)
exten => s,4,Wait(1)
exten => s,5,Background(custom/ivr-incoming)

exten => 1,1,Background(pls-wait-connect-call)
exten => 1,2,Dial(${RINGPHONENUMBERS},20,r)
exten => 1,3,Voicemail,u${VMBOX}
exten => 1,4,Hangup

Running * 1.0.5. The calling party hears the please wait while I
connect your call, but does not hear any ringing. I tried inserting
exten => 1,1,Ringing but that does not work either. 

The same call flow from the pstn DOES generate ringback:
[fromPSTN]
exten => s,1,DigitTimeout(2)
exten => s,2,ResponseTimeout(10)
exten => s,3,Wait(1)
exten => s,4,Background(custom/ivr-greeting)

exten => 1,1,Background(pls-wait-connect-call)
exten => 1,2,Dial(${RINGPHONENUMBERS},15,r)
exten => 1,3,Voicemail,u${VMBOX}
exten => 1,4,Hangup

Any thoughts.

Some more info. Using this exact call flow, ringback works for PSTN
callers over WIldcard, IAX Callers over VP Connect, but NOT IAX
callers over LiveVoip. Could this possibly be a bug with their "new"
patch?


On Wed, 26 Jan 2005 11:43:59 -0500, Brian Dingman <bdingman@gmail.com>
wrote:
> Here is the call flow:
> [ivr-incoming]
> exten => s,1,LookupCIDName
> exten => s,2,DigitTimeout(2)
> exten => s,3,ResponseTimeout(10)
> exten => s,4,Wait(1)
> exten => s,5,Background(custom/ivr-incoming)
> 
> exten => 1,1,Background(pls-wait-connect-call)
> exten => 1,2,Dial(${RINGPHONENUMBERS},20,r)
> exten => 1,3,Voicemail,u${VMBOX}
> exten => 1,4,Hangup
> 
> Running * 1.0.5. The calling party hears the please wait while I
> connect your call, but does not hear any ringing. I tried inserting
> exten => 1,1,Ringing but that does not work either.
> 
> The same call flow from the pstn DOES generate ringback:
> [fromPSTN]
> exten => s,1,DigitTimeout(2)
> exten => s,2,ResponseTimeout(10)
> exten => s,3,Wait(1)
> exten => s,4,Background(custom/ivr-greeting)
> 
> exten => 1,1,Background(pls-wait-connect-call)
> exten => 1,2,Dial(${RINGPHONENUMBERS},15,r)
> exten => 1,3,Voicemail,u${VMBOX}
> exten => 1,4,Hangup
> 
> Any thoughts.
>

Finally got a reply from LV support. Not what I was hoping for.
Hopefully they will file a bug with Digium since they investigated the
issue.... not holding my breath.

Since this is such basic * functionality that they can't seem to
accomplish I would think twice before aquiring DID's from them.

---------------------------------------- LiveVoip Support

Our people have looked into this matter over the past few days. They tell me
that it is a problem with Asterisk.
We are not going to be able to help you with this. If you would like a
refund so that you can migrate to another
service provider we will be happy to do so. With each rev. of Asterisk more
and more improvements are made.
At some point these issues may resolve but, for the time being it is not a
problem we can help you with.


On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier <lists@futuresync.com>
wrote:
> I just got a couple of numbers (activated Friday) from livevoip, I am
having
> similar issues.
> 
> When you call the number, I get ring back, but as soon as IVR picks up, I
> should here "extensioni" I don't hear that but then I dial an
extension
> number and there is no ring back.  I don't have this issue from other
voip
> providers.
> 
> Steve
> 
>

On Fri, 04 Mar 2005 11:46:27 -0700
  Paul Fielding <paul@fielding.ca> wrote:
> Hmmm.....  My server is currently set to let the line 
>ring for 20 seconds, ringing several extensions 
>internally.  (I do not answer the line, it just rings the 
>extensions).  If I don't pick up after 20 seconds it then 
>answers the line and sends to voicemail or to an 
>auto-attendant, depending on the situation.
> 
> Ringback seems to be working for me, I hear ringing on 
>the calling end... *shrug*.
> 
> Paul
Ok,I have to retract my last statement and give an update. 
It has been a while since I had played with the DID I have 
from them.

It is not an issue before the * box picks up. I set my 
incoming context to ring my VoIP phone for 20 seconds 
directly with using the IVR system and I had the ringing.

But when I restored it to no background on hold music and 
issued a dial command of Dial(SIP/2001,15,r) instead of 
Dial(SIP/2001,15,m), after the IVR plays its intro, I got 
no ringing on the calling end. Just dead air from 
LiveVoIP.

I then used this same test context by dialing in through a 
VP Connect account and after the initial greeting and 
moving to the Dial command, I got the ringing on the the 
calling end.

Sorry for the incorrect info the first time, it had just 
been quite a while since I had played with the Live 
account.


Robert

> 
> ----- Original Message ----- From: "Robert Webb" 
><asterisk@ropeguru.com>
> To: "Asterisk Users Mailing List - Non-Commercial 
>Discussion" <asterisk-users@lists.digium.com>; 
><paul@fielding.ca>
> Sent: Friday, March 04, 2005 11:42 AM
> Subject: Re: [Asterisk-Users] Re: No ringback over IAX - 
>LiveVoip
> 
> 
>>
>> On Fri, 04 Mar 2005 11:35:55 -0700
>>  Paul Fielding <paul@fielding.ca> wrote:
>>> Ok, time for me to ask my own newbie question.   :) I've 
>>>done some 
>>> digging on ringback, and if I'm understanding it 
>>>correctly, it's the ring 
>>> tone that the caller hears when dialing another person.
>>>
>>> What exactly is it that people are finding now working 
>>>with LiveVoip? 
>>> Everyone says 'ringback isn't working', but
nobody's
>>>really explained 
>>> exactly what's happening. At least not that I've been 
>>>able to find.
>>>
>>> I have a DID with them, and it works just fine. Dialing 
>>>out works fine, 
>>> when people call in it works fine.
>>>
>>> I'm interested in knowing what it is that isn't working, 
>>>and if I can 
>>> re-create it on my system...
>>>
>>> regards,
>>>
>>> Paul
>>
>> Setup your * box to not answer the call right away. 
>>Allow for say 5 
>> seconds of ringing. Then call into it on one of your 
>>DID's. From the 
>> calling end all you will get is dead air. No ringing.
>>
>> At least this is the issue I am having..
>> 
> 
>

Hi

I am running Asterisk CVS-v1-0-03/04/05-18:54:35 and I see to get the
error stated above.

My setup is ser which takes in a call and rewrites to asterisk.

ser.cfg :

 rewriteuri("sip:10@a.b.c.d:5090");


This takes call to asterisk

sip.conf

[general]
autocreatepeer=yes
port=5090
defaultexpirey=3600
register => user:pass@sip.d.e.f/10
context=sip

This creates a peer with the sip server.

extensions.conf

[sip]

exten => 10,1,SetCIDName(Test Line)
exten => 10,2,Dial(SIP/test@sip.d.e.f)


So from the command line in asterisk I try

dial 10@sip

this sends details to ser, ser picks them up, but asterisk shows this , I
have obvioulsy missed something very simple here

tks

Iqbal