Displaying 20 results from an estimated 8000 matches similar to: "Asterisk set-up for LCR"
2005 Aug 16
8
Asterisk and LCR
Hello,
How do you guys implement LCR in Asterisk?
Thanks,
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 Mar 13
2
How can I eveluate trailing numbers in extensions.conf?
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote:
> Hi,
>
> this
2006 Jan 23
1
How to set-up LCR
How to set-up LCR ?
a. which companies can be used with LCR?
b. how to set-up & maintain LCR?
c. multiple connection to one gateway?
Example:
+886223456789 could be reachable via
a. ENUM free
b. Dundi free
c. Voipstunt free
d. Voipbuster free
e. Nufone $
f. Voipstunt $
g. others with 4 concurrent connections $$
h. others with 3 concurrent connections $$
I am looking
2004 Jun 07
2
slightly OT: VoIP more expensive than Call-By-Call
Hi,
is it just me, or are the VoIP providers for Germany more expensive than
going via call-by-call?
sipgate.de lists a price of 1.76ct a minute, a couple of call-by-calls
are listed at 1.3ct-1.5ct a minute.
Did anyone else thought this to be strange?
What about other countries? Same thing?!
It seems to me, VoIP providers only make sense for headquarter to
subsidiary, but not for regular
2008 Feb 12
3
LCR in Asterisk
Hi all,
I am planning to implement LCR routing on my already running asterisk
server. Uptill now i have found out that asterisk has no support for lcr, i
have to do something about it myself, for example using the AGI. Im looking
for ideas here. Whats the best way to start implementing lcr in asterisk.
Should i use agi and start implementing my own lcr script or is there any
plugin available which
2006 Jan 24
8
UK Provider
Hi
Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account.
www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations.
Many Thanks
Scott
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
2006 Jun 16
2
MOS Scores and LCR
Is there any tool that can do LCR for Asterisk but also take into
account MOS scores?
Is it possible to automatically generate MOS scores on random "calls"
so as to keep an updated database on a per provider, per destination,
per time-of-day score? Hopefully, with that information we can create
a better LCR module or script?
Thanks,
Daniel
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device
192.168.3.99 (same ports)
other direction is totally open.
I
2009 Nov 22
1
transferring SIP call: no voice
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and Teliax.
CLI on A looks right:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
==
2009 Apr 26
1
sipgate doesn't work with sipgate anymore
Hi,
have some problem with incoming calls from sipgate. This was working in
1.4 but in 1.6 I get a 401 Unauthorized :-(.
Sipgate has mentioned that I have to change the type to friend, but it
is already friend, so what's wrong?
Kind regards,
Michael
Here is the sip.conf:
[sipgate_out]
type=friend
nat=yes
username=1234567
fromuser=1234567
fromdomain=sipgate.de
secret=secret
host=sipgate.de
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 Dec 23
2
Incoming calls from Sipgate go through the wrong peer
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it seems to pick the
wrong peer from sip.conf which sends the call into the wrong context and
it fails because there is no extension in that context to match the
register.
Using the config's below, if I dial the DiD on account 2222222, it works
fine - picks peer 2222222
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
2005 Feb 08
2
Asterisk and Sipgate problem...
Hello all. I'm having an odd problem getting * and sipgate to work
together. From Sipgate support I have gotten this repsonse to my query:
=====
Your Asterisk is registering incorrectly with our servers. It registers
like this: sip:s@217.XXX.XXX.XXX:5076
The "s" should be your SIP ID. Anything else is rejected. I don't know
where you can find this setting, but from our
2006 Feb 25
2
sipgate.de question
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n)
I looked at the sip debug stuff, and all I
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
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
ping times are fine as well:
[root at freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57