[Asterisk-Users] carrier connection options

2004-09-06 Thread Peter Boot
Can anyone tell me if there is an alternative option to connecting Asterisk
as a SIP server to a carrier using T1/E1 lines or a carrier that will
terminate a routed SIP call ? 

I am looking for a more cost effective solution that will avoid the setup
and incremental cost of VOIP gateways and T1/E1 lines as the need for more
and more concurrent calls escalates.
Thanks in advance.

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RE: [Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?

2004-06-24 Thread Peter Boot
I had the same problem when using a Grandstream 486 I solved it by using the
nat=yes config option

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Andrew Yager
Sent: Friday, June 25, 2004 3:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?

Hi,

I'm confused as anything by this bug. I'm hoping that it is 
just something screwy in my config.

I have 1 Cisco 7960 and several Grandstream BT101  102's, 
and a Digium TDM31B.

I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of 
both Asterisk and the Zaptel driver on Fedora Cora 1.

When I make an outgoing call on the Cisco phone, everything 
works fine. 
I'm connected, and it all sounds hunky dory.

My Grandstream phones talk quite nicely to Asterisk. I can 
receive incoming calls and have them forwarded to my phone, 
and I can dial internal extensions without a problem. 
However, whenever I attempt to make an outgoing call, the 
outgoing number rings, but no audio is ever sent to the 
Grandstreams, even when the phone is answered. If I put an r 
in the dial plan, the GrandStream does generate the ringing tone. 
When an m is set, no audio is transmitted to the phone. The 
person who answers the call hears absolutely nothing at all.

The Grandstream phones can talk to each other without a problem.

It seems that the bug is being generated between the 
Grandstream phones and the Zap card, but only on outgoing calls.

To add to the confusion, if I phone one of the FXS ports 
connected to our hard fax, it rings, answers and everything 
works just fine.

My zapata.conf is presently:

[channels]
context=incoming
signalling=fxs_ls
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=no
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes ; have tried changing this to yes and 800 - 
no difference rxgain=0.0 txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=no ; previously had this at yes, but when changed 
it to no to test
busycount=4
musiconhold=default
faxdetect=incoming
channel = 1

Any help or suggestions on what to try or where to go would 
be appreciated.

Andrew

_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
ph: (02) 9945 2567 fax: (02) 9945 2566
mob: 0405 15 2568
http://www.rwts.com.au/
_

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[Asterisk-Users] nat=yes

2004-06-07 Thread Peter Boot
I am trying to use asterisk as a gateway between SER and the PSTN.

Should the nat=yes config work with these sip.conf settings ?asterisk is
trying to send it's response 
back to the private IP.


[general] 
context=OUTGOING 
autocreatepeer=yes 

[Provider] 
type=friend 
username=X 
secret=X 
host=x.FakeProvider.com 
nat=yes


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[Asterisk-Users] X100P to hardware PBX

2004-06-02 Thread Peter Boot
I have asterisk successfully dialing out using a X100P over a normal
analogue PSTN line. But when I try to dial out over an analogue line that
goes via a hardware PBX the call asterisk does not dial. Is there a
configuration change I should make ? I am thinking of something like not
wating for dial tone.

From my extensions.conf
[outgoing]
exten = _X.,1,Dial,Zap/1/${EXTEN}


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