Displaying 20 results from an estimated 400 matches similar to: "R: Configure asterisk for outgoing.. need authuser parameter?"
2004 May 18
0
No luck using asterisk as proxy...
Still no luck using asterisk as a proxy.
48 hours solid working on this. I'm beginning to think asterisk isn't
going
to be compatible with the provider I'm using :(
Has anyone got *any* clues as to what can cause this message? It's
definately
provider specific (voiptalk works, pipecall doesn't) but confusingly
seems to
be caused by something in the client phone app.
I
2004 May 18
1
Configure asterisk for outgoing.. need authuser parameter?
Hi,
I have access to two providers. On one of them the authuser is the same as
the username, so outgoing works. On the other one I can only get
incoming -
what ever combination I try for outgoing I get an error. The register
command
has the ability to specify both usernames (which is why incoming works) but
outgoing doesn't seem to, and without that I'm stuck.
They are defined as:
2004 Jul 23
0
Pipecall problem
I have been a reseller & subscriber of pipecall since they started,
however I am really struggling to get pipecall to work for outbound or
inbound calls. I get errors that the registration has timed out.
I have tried many variations of the register command
register => 0845xxxxxxx@sipproxy.pipecall.com/1000
register => sipxxxxxxxxx:xxxxxxxxxx@sipproxy.pipecall.com/1000
2004 Jul 07
1
UDP Ports scan on firewall
I'm using Asterisk to registry several DDI's to a sip proxy
(pipecall.com). Everything works fine apart from several times a day my
firewall (zywall70) reports a UDP port scan attack from the pipecall sip
proxy. I can't seem to work out why this should be. All I could think
was that the sip registry was expiring and causing some strange probing
from the proxy, is it possible to alter
2004 Sep 02
1
Any UK PipeCall/PipeMedia users?
Has anyone out there used the PipeMedia/PipeCall PSTN gateway?
Anything good/bad to say about it?
I'm considering using them for a new customer. They seem to have good rates,
good provisioning tools and (better still) give commission on usage to
dealers.
--
David Gurr
Congruity Ltd. Fax: 0871 661 1756
Hemel Hempstead
UK
2004 Jun 16
1
VOIPTalk silver service
There was some discussion on this list recently about the voiptalk silver
service. I've just had an e-mail from them saying that the price has been
reduced to 2.99 per month. However, they still only provide an 0870 number
whereas pipecall provide a local call rate 0845 number in the fee.
Chris
2006 Apr 17
4
Looking for a good VoIP Provider in the UK-
Any recommendations for a VoIP provider in the UK?
I have a few guys in a field office in the UK with SIP phones and a VPN
tunnel back to a working Asterisk setup in the US. The Asterisk setup
has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US
offices, so they can call vendors, customers etc in the US at local
rates. I'd like to get the same thing for the UK, so that UK
2004 Sep 23
0
Duplicated INVITE in SIP session?
Hello.
I'm trying to use Asterisk in combination with SER, to make the
routing proccess to my PSTN-Gateways. I made a simple test defining some
extension in my extension.conf, when i made a call my SER (SIP) Server
forward the call to Asterisk, this proccess is ok, but when the call is
answered i see an INVITE going out from Asterisk to my SER Server, this
invite is then passed to my
2007 Jan 15
1
I have to register asterisk/sip with a sipproxy that does not support authentication?
I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request?
# sip.conf
[general]
insecure=very
permit=207.148.115.10/255.255.255.0
[myproxy]
type=friend
host=217.118.115.10
context=from-sip
# Logging:
<--- Reliably Transmitting (NAT) to 207.148.115.10:5060 --->
SIP/2.0 407 Proxy
2005 Jul 14
1
PSTN to SIP gateway
I've been looking through the examples and docs, but haven't yet quite
figured out how to do what I want.
We've got a T1 coming in carrying a block of telephone numbers,
terminating on an Asterisk box. Any call that comes in needs to get
sent to a SIP proxy, with a particular extension format:
*ANI*DNIS*@sipproxy.address
The closest I can see to do this with the Dial() command is:
2009 Dec 27
2
Call ends when picked up
Hello list.
My phone rings, I pick up, and the conversation is terminated. Every
time.
The setup :
Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server
--> ITSP
Could it be the SIP proxy of my Endian firewall ??
I have 4 accounts on the Grandstream which listen on port 5060 --> 5063.
They have a proxy defined namely my Endian firewall.
On this SIPproxy I have a
2009 May 15
1
Spiral SIP Request problem
Hello,
I am using OpenSIPS to register all the users and planning to use asterisk
for Auto Attendant, Queues, Voicemail and Conference Bridge.
I have a scenario where the signaling does not happen properly:
1) A user from Opensips dials an extension 7000 which is an
auto-attendant extension. The call is routed to asterisk to play the auto
attendant messages like Welcome and Dial the
2005 Mar 17
3
Channel name (and substring)
How do I get the bit like "IAX2/white_phone" in extensions.conf eg from
pre-defined variables or variants thereof ?
What I *do* get is strings like this "IAX2/white_phone@white_phone-4"
from ${CHANNEL}, but that's the full channel name.
What am I missing here ?
Thanks,
Thomas
2007 Jan 11
2
Voicemail IMAP
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email server.
We are using a non-standards based groupware server called FirstClass.
The server has some built in support for IMAP.
My problem seems to be that the authuser flag is not supported.
When I use mtest in the imap toolkit to connect to
2006 Dec 27
8
1.4.0, IMAP and Dovecot
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
2005 Mar 05
7
BroadVoice configuration changes for Outbound
Today, We have added INVITE Authentication. This seems to bring a large
amount of problems to people in the way since they can't make outbound
calls. Here's what needs to be done. You need to add three variables to
your peers or friends, username, authuser, and secret.
username=<phonenumber>
authuser=<phonenumber>
secret=<registration password>
Dan
2011 Jun 29
5
[Urgent] Email Retrieval from remote servers doesn't use Dovecot
------------------------
Dovecot Version:
------------------------
2.0.13
------------------------
Output of "dovecot -n":
------------------------
# 2.0.13: /usr/local/etc/dovecot/dovecot.conf
# OS: Linux 2.6.35-28-generic x86_64 Ubuntu 10.10 ext4
mail_location = maildir:/home/%u/Maildir
passdb {
args = %s
driver = pam
}
protocols = imap pop3
ssl = no
userdb {
driver = passwd
}
2004 Dec 10
0
Confused about proxying and NAT, and seeking guidance
I think I have got * worked out as far as getting users on a small
private network talking with each other, but when it comes to the bigger
picture about talking between private networks connected by the Internet
then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc.
Before I start let me make it clear that I am not looking to drop out
onto the public telco network anywhere, not at
2007 Jul 18
1
Asterisk Voicemail Imap Storage with MS Exchange
Hi,
did anybody manage to configure Asterisk (1.4.8) with imap voicemail
storage together with an Microsoft Exchange Server (2003)?
I can connect my asterisk to a local dovecot imap server without problems.
But if I change the settings to our exchange server I can?t connect...
Here is my config:
imapserver=exchangeserver
imapport=143
expungeonhangup=no
imapfolder=INBOX
imapflags=notls
2005 Sep 05
2
USING TWO ACCOUNTS WITH BROADVOICE
Hi,
I have two accounts with broadvoice.
Now, I want to be able to distinguish between them.
I though that this would be simple by adding "/EXTEN" at the end of the
register statement. For example:
register => num1:pass@sip.broadvoice.com/1000
Unfortunately, this is not working.
When I call into my box I hear busy tone.
My config looks like this:
[root@voip asterisk]# cat sip.conf