My DID with Telasip is disconnected and my Asterisk box wont
register with them. Anyone else having problems with them?
Jeff
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 14, 2005 12:00 PM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users Digest, Vol 13, Issue 98
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I have asterisk with two zap channels which are analog ports off a T1. They
each have a inward DID number If they are used for outgoing they show the T1
main number not the DID's number. Is there any way to send caller ID of the
inward DID number not the main number
Jeff
Message: 9
Date: Sun, 3 Apr 2005 23:52:39 -0500
From: * KAPIL * [EMAIL PROTECTED]
Subject: [Asterisk-Users] [EMAIL PROTECTED] Question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Greetings!
This is my first post to the list...and
Message: 16
Date: Fri, 25 Mar 2005 01:06:21 -0700
From: JD [EMAIL PROTECTED]
Subject: [Asterisk-Users]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I'm
Vicky Shrestha Wrote
Message: 8
Date: Wed, 23 Mar 2005 19:16:49 +0545
From: Vicky Shrestha [EMAIL PROTECTED]
Subject: [Asterisk-Users] Broadvoice alternatives
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Here is my BV info which works for outbound
[BV]
username=phone-number
user=phone-number
type=peer
secret=PW
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=phone-number
fromdomain=sip.broadvoice.com
dtmfmode=inband
authuser=phone-number
___
I had my [EMAIL PROTECTED] server working fine SPA-8841
SPA-2100. It was on an open IP no fire wall. I moved the server
behind the firewall. Now the phones will not dial out. The phones
can be called from a DID or calling to the main POTS number and dialing the extension.
However neither
I had to redue the zapata.conf
Commented it out ans added, Also changed default Zap g0 to Zap 1 (deleted
Zap g0)
Where did you find [EMAIL PROTECTED] .4 all I see is 0.3
Jeff
[channels]
language=en
;context=inbound-analog
;context=default
context=from-pstn
signalling=fxs_ks
usecallerid=yes
? [EMAIL PROTECTED] V 0.4
It was released yesterday, I believe.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jeff R Glassman
Sent: Friday, January 28, 2005 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk
-Commercial Discussion'
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4
What is new in .4?
Thanks
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff R
Glassman
Sent: Friday, January 28, 2005 4:08 PM
To: Asterisk Users Mailing
: FAQ missing info? [EMAIL PROTECTED] V 0.4
Yes, now runs an updated version, also has MeetMe2 which is worth a
donation all on it's own.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff R
Glassman
Sent: Friday, January 28, 2005 4:08 PM
I has Asterisk up and running on my IP address.
I put a Linksys router in front of it and forward the following ports
22 TCP
5060 UDP
1-2 UDP
80 Both
None of my x ten phones work. They register but I get an message
Authentication Required
[202]
username=202
type=friend
I also edited the Zapata.conf file I did not change the zaptel.conf, what did
you change in it
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David Shaw
Sent: Thursday, January 27, 2005 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial
I have a DID from Livevoip. It is a low usage DID. I have had no problem
with support. I get a reply back within the hour on the couple time I had a
problem.
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Dingman
Sent: Thursday, January 27,
I am running [EMAIL PROTECTED]
Voicemail works fine but does not email out the voicemail attachments. Any
suggestion?
---
Voicemail.conf
[general]
#include vm_general.inc
#include vm_email.inc
[default]
201 = {password},Jeff G Laptop,[EMAIL
Just to let everyone using [EMAIL PROTECTED] know that my livevoip DID now works
without any changes to asterisk!
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Eissler
Sent: Wednesday, January 26, 2005 10:00 AM
To: Asterisk Users Mailing List -
Just a thought SPA-2100 will give you 2 ports and a lan port per ata.
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bill
Lattner
Sent: Wednesday, January 26, 2005 6:25 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] setup questions-
I have a PCPHONELINE SIS phone set it up to asterisk
Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes
every time)
Got SIP response 481 Call Leg/Transaction Does Not Exist back from
24.172.221.22(24.172.221.22 is my phones IP)
Anyone have an idea what the problem is?
I have an X100P clone hocked up to an analog line of my PRI. I can use it
to dial out.
but when I call the extension it answers and says GOODBY
I have a Livevoip DID which successfuly rings to ext 202
I am using [EMAIL PROTECTED] and through the AMP inface the line should ring to
ext 202
According to Ed Guy at Bellster
The most specific routes takes precedence. For example, if you are
calling 1-212-555-1212 first routes for 1-212-555 are checked, then
1-212, then 1 until a non-congested route is found. (The searching is
actually a bit more general -- matching is done on a per
I am running * with a single X100P (Clone) card. I would like to add 1 - 2
more. Is there a limit to the number of X100P cards other than the PCI
slos on the motherboard?
Jeff R Glassman
VP
Columbus Green Cabs, Inc.
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Same here!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Levenson
Sent: Friday, January 21, 2005 6:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bellster - IAX-based interchange -- lets
I
beleive both are locked into a VOIP carrier (Vontage?)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of James H.
ThompsonSent: Wednesday, January 12, 2005 11:54 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
I am planning to have our ISP (RR Business) open up the necessary ports so
our Asterisk box which is behind a firewall can communicate with Sip phones
outside the firewall.
From searching the list archive I have come up with the following list
22 for SSH Should this be TCP, UDP or Both?
Did you ever have success copying your configs to the Xorcom box?
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Monday, January 03, 2005 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
What ports need to be Fowarded?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wilson
Pickett
Sent: Tuesday, January 11, 2005 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP, * and clients behind NAT
Try http://knopsterisk.com/
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Shahed
Moolji
Sent: Wednesday, January 05, 2005 11:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Bootable Asterisk CD ?
Hi All,
A while ago, I saw some
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