Displaying 20 results from an estimated 4000 matches similar to: "multiple sip accounts from same sip registrar"
2005 May 16
4
Asterisk@home 1.0 + Sipgate UK/SIP Provider
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk
2004 May 19
1
Strange Sip (FWD, SipGate and such) problem
Hi all
I use sipgate and FWD but seem not to get it going. I do not have NAT on
the asterisk box (static ip).
The asterisk box has 2 network interfaces. One internal and one external.
Now when I make an call to a FWD or SipGate number all I get is
-- Executing NoOp("SIP/113-6d2e", "") in new stack
-- Executing Goto("SIP/113-6d2e",
2004 Jun 27
5
Optipoint 400 Standard Sip
Hi everybody,
I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk.
It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers.
The Optipoint shows "no Server..." (Registrar?) in Display.
Sip debug shows no unusual (to me) Messages.
Sip show
2004 May 08
3
asterisk with german SIPGATE ?
hi
anybody running with german SIPGATE?
my configuration don't works :-(
regards
thorsten@gehrig.de
2010 Feb 05
0
Sipgate.co.uk on Asterisk 1.6.2.2
Hi all,
I have been running Asterisk for years (CVS-HEAD on 2005-08-24) with no
problems save a failed harddrive. I have decided to build a new box and have
Asterisk 1.6.2.2 playing nicely with mISDN after lots of changes to dialplan
syntax etc. I am struggling with SIP trunks to sipgate.co.uk and
dualtalk.com. Does anyone have a working examples?
When I make an outgoing call I get...
[Feb 5
2009 Feb 24
2
Multiple SIPGate accounts.
Hi all,
I have two sipgate accounts (numbers), if I have both accounts register only
one will work for incoming calls (which is all i'm interested in). However
if I disable either account the other account will work perfectly. Am I
missing something obvious?
Thanks in advance,
Ray.
Excerpts from sip.conf -
[general]
8<---- SNIP! ---->8
Register => 1212121:aaaaaaaa at
2006 Oct 25
2
Call is not coming through sipgate.co.uk+Asterisk
Hi,
I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx.
I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I
2009 Dec 15
3
Best way ro run 2 or more asterisk servers?
Hello List.
I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one.
I would like to run different scenarios:
1. Have one of the boxes at a different location outside the LAN and have them communicate.
2.
2005 Aug 04
1
Getting asterisk to work with callthroughs?
Hi,
Firstly, what I'm trying to do is:
* Get asterisk to pick up a SIP call via a DID
* Prompt the user
* When the user puts in a number, go to IAX.conf and route it according to
what I've specified there, i.e Least Cost Routing, etc.
I've set-up something similar to what I've found online, but it doesn't
work! Asterisk doesn't pick up the call at all..... :(
The files
2005 Oct 10
1
Incoming SIP getting in, but not ringing.
Hi all.
Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)
Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great from extension 8 on phones. However some
more friends/contacts have started using SipGate also, and
2004 Sep 26
6
SIP Registration Timeout, No FW
Hi people,
My asterisk wont register with any sip providers, I have tried three
different but they all end up with:
Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout:
Registration for 'whatever@provider.tld' timed out, trying again
There is no firewall and my server has a public IP. Could this be a Asterisk
problem?
-Fredrik vK
2014 Dec 30
3
status - Unmonitored, how to change it
How to change status of peers "Unmonitored" to monitored?
Home users showing "Unmonitored" some display timing.
Name/Username Host Mask Port Status
zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored
clinic_server (null) (D) 255.255.255.255 0 Unmonitored
voip
2004 Dec 06
1
iax2 nativ bridge question?
hallo all,
i would like to know, as i would suspect, nativ bridiging should work also,
if only one iax party is behind an nat router and the other has a public
ip. when i connect to iax clients, which have both pubic ip's nativ
bridging is working. if one of the clients is behind an nat, the iax2
channels always get routed through the asterisk server (latest stable
version from cvs) ?? i
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all.
The asterisk setup is working fine, receiving calls via broadvoice "initially". ?
When call comes in via broadvoice number, asterisk picks it up and routes
correctly, as long as the call came in within ~2 min from the previous one.
In other words, as long as a call comes in within ~2 min since the previous one,
asterisk will answer the call. However, if the call comes in
2004 Jun 15
3
Grandstreams randomly go busy with Asterisk?
I've searched the lists but I didn't find anything exactly like this.
I have two Grandstream BT101 phones connected to an Asterisk.
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy. Dialing that
phone gives:
-- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack
2004 Jul 08
2
Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The
asterisk box is on a WAN connection on the other end of a DS3, the
phones connect fine to the Asterisk server as you can see from the
output of show sip peers below.
tp3/tp3 <firewall-ip> D N 255.255.255.255 60665
Unmonitored
tp2/tp2 <firewall-ip> D N
2005 Mar 21
1
Net2Phone / Vonage
I can cut and paste the log file from a reload right now, and provide
you with the other information when I get home after work:
tmp*CLI> sip debug
SIP Debugging Enabled
tmp*CLI> reload
Mar 21 14:52:42 NOTICE[23231]: indications.c:397
ast_unregister_indication_country: Removed default indication country 'us'
11 headers, 0 lines
Reliably Transmitting:
REGISTER
2005 Aug 02
2
asterisk@home newbie extensions always busy
hi list,
I'm running a newly installed asterisk@home an i registered two soft
phone. both soft phone are registered
8901/8901 x.x.x.x D 255.255.255.255 50710 Unmonitored
8900/8900 y.y.y.y D 255.255.255.255 6281
Unmonitored
but when I call one another, they are always busy and directed to its
voicemail
Sorry, if this was posted before
TIA
2013 Sep 18
2
sipgate outgoing calls
Hello
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line
-- Called 01179248615 at sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1'
--
2009 Oct 18
1
Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?
I'm trying to setup sipgate on 1.6.1. Following the instructions on the
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser= ;;SIP-ID
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com