asterisk users - Apr 2004 - Strange T1 Problem

When people call into my * box over the T1 interface, they get no ring
tone. It rings the SIP phone and when the SIP user picks up, both
parties can hear each other ok, its just the PSTN user calling in hears
no ring. What could be causing this?

I tried setting immediate to yes in zapata.conf, but that causes my DNIS
and CallerID to stop being available.

T100P with E & M Wink start signaling, all 24 channels are inbound
channels (no channel bank or anything like that) to SIP ATAs. The ATA is
sending a 180 Ringing reply to the invite, but still no ring. Same
symptoms with different vendor ATA devices.

On Thu, 2004-04-15 at 17:18, Mike Machado wrote:
> When people call into my * box over the T1 interface, they get no ring
> tone. It rings the SIP phone and when the SIP user picks up, both
> parties can hear each other ok, its just the PSTN user calling in hears
> no ring. What could be causing this?
> 
> I tried setting immediate to yes in zapata.conf, but that causes my DNIS
> and CallerID to stop being available.
> 
> T100P with E & M Wink start signaling, all 24 channels are inbound
> channels (no channel bank or anything like that) to SIP ATAs. The ATA is
> sending a 180 Ringing reply to the invite, but still no ring. Same
> symptoms with different vendor ATA devices.
Explicitly answer the line. If that doesn't handle inband audio, there
is a r flag to dial. This was discussed very recently. 
-- 
Steven Critchfield  <critch@basesys.com>

Fixed in CVS STABLE around 2pm CDT today.  It's been fixed in CVS HEAD 
for a while.

Mike Machado wrote:
> When people call into my * box over the T1 interface, they get no ring
> tone. It rings the SIP phone and when the SIP user picks up, both
> parties can hear each other ok, its just the PSTN user calling in hears
> no ring. What could be causing this?

Mike Machado wrote:
>cvs HEAD did infact fix the ringing problem. Thanks Eric!
>
>I have another question for all you T1 buffs out there. The T1 I am
>working with goes into our local phone switch (Excel switch). Currently
>we are using E & M Wink signaling. The problem is we cannot set callerid
>on the outbound side. My minimal understanding is that if we had a PRI,
>I could set the callerID. Unfortunately PRI is one signaling type they
>cannot do (not have expensive PRI card in switch).
>
>So, my question is what other signaling types CAN I set the callerID
>outbound?
>
>My local switch techs cannot seem to answer that question. They just
>always use E & M for everything. But if I can ask them to specifically
>try a certain signaling type (such as Feature Group D) or one of the
>others in the t100p supported list, I could probably get them to change
>the signaling type on my trunk. Do any signaling types other than PRI
>support passing outbound callerID?
>  
>
You can usually get CLI on an E&M robbed bit T1 by configuring it right. 
Instead of just sending you the DNIS as a string of DTMF they usually 
send *<cli>*<dnis>*. The DNIS and CLI may be swapped, and there may
be
less than 3 *s in the string - wonderful consistency, eh? :-\

Regards,
Steve

Mike Machado wrote:
> cvs HEAD did infact fix the ringing problem. Thanks Eric!
As I said, CVS STABLE also has the fix as of this afternoon.

Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
   Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well.  It will just ring
forever.
Cheers,
Willy

----- Original Message Follows -----
> 
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
> 
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently. 
> 
> This must be a different problem, because neither of those
> solutions worked.
> 
> 
> 
> zapata.conf sends call to fixup context:
> 
> 
> [fixup]
> 
> ; Receive call as *<calling>*<called>
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
> 
> 
> [default]
> 
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
> 
> 
> user1's phone rings, but no ring from PSTN caller. user1
> picks up, both can talk ok.
> 
> 
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
Willy Wouters
ypOne Publishing

Hello,
>From: "Joe Dennick" <joe@dennick.net>
>Subject: RE: [Asterisk-Users] Strange T1 Problem
>Date: Fri, 16 Apr 2004 07:44:17 -0500
>
>Can one use a pipe '|' for the Dial application the same way that
one
>would use a comma ','?
<snipped>
>I know this one works, but what I don't know is if it will also work
>using pipes in place of the commas.
>
>Joe
>
Yes, You can use '|' for the dial application. In fact even if you use 
comma(,) in your extensions.conf,  Asterisk replaces it with '|'  when
it
builds the dial plan.  See the following entry in extensions.conf:

[test]
exten => 1234,1,Dial(SIP/1234,20,r)
exten => 1234,2,Voicemail(u1234)

and the dialplan for this is:

* CLI> show dialplan test
[ Context 'test' created by 'pbx_config' ]
  '1234' =>         1. Dial(SIP/1234|20|r)                
[pbx_config]
                           2. Voicemail(u1234)                    
[pbx_config]


Regards, Girish

_________________________________________________________________
Easiest Money Transfer to India. Send Money To 6000 Indian Towns. 
http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!

> >
> You can usually get CLI on an E&M robbed bit T1 by configuring it
right.
> Instead of just sending you the DNIS as a string of DTMF they usually 
> send *<cli>*<dnis>*. The DNIS and CLI may be swapped, and there
may be
> less than 3 *s in the string - wonderful consistency, eh? :-\
I am getting CallerID and DNIS on the inbound calls. What I really need
is to be able to set callerID on outbound calls. I am trying to set the
callerid using SetCIDNum just before using Dial on a zap channel, but it
looks like the switch guys have it set to always stamp the same callerID
on the my outbound calls no matter what I put in SetCIDNum or what
channel on the T1 I use. Is this a misconfiguration of the switch or a
limitation of the signaling protocol? If its the switch, can you give me
any pointers as to what I could ask them to look for, or if its the
protocol, do you know any other signaling protocol that lets me set
outbound callerID (besides PRI)?