Displaying 20 results from an estimated 300 matches similar to: "chan_sip doesn't authenticate on INVITE from a Dial() command"
2020 Oct 25
0
chan_sip doesn't authenticate on INVITE from a Dial() command
On Sunday 25 October 2020 at 16:27:00, Antony Stone wrote:
> Hi.
>
> I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and
> for some reason it's simply not doing it.
I've made a bit of progress - I can now get it to authenticate, although it's
still not dialling on to the correct number.
> I've even resorted to reading the source code
2008 Jan 31
1
Dropped calls
I have a very serious problem with calls between PAP2-NA and a TDM2400 (8
FXO). Almost every call dropped after between 20 and 30 seconds with
conversation.
I disable the sound card, serial and other things on my server, but the
problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA,
but nothing.
Here a piece of my log:
[Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up
2020 Oct 27
1
Bug in Dial() string processing
Hi.
I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at
least).
According to the documentation in channels/chan_sip.c the Dial() string syntax
is:
* SIP/devicename
* or SIP/username at domain (SIP uri)
* or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
* or SIP/devicename/extension
* or SIP/devicename/extension/IPorHost
* or
2012 Oct 31
2
Asterisk and OpenLDAP
Hello guys,
i would like to implement authentication for my sip extension with an
openldap server.
Following this guide
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
i see a template named [sip] to map the information of sip peers into ldap.
But i'm not interested to create a template, i would only authenticate
sip extensions using username
2004 Jan 30
2
Can Asterisk act like a normal sip phone?
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Hash: SHA1
Hello everyone,
I'm relatively new to the subject - so pleace don't punish me for
idiotic questions. ;-)
Can Asterisk act like a normal Sip phone and e.g. connect to another
sip-gateway? Background: There is a new german company at:
http://www.sipgate.de (sorry German only page)
They offer a a gateway between a real telephone number and
2005 Oct 13
1
AGI Variable problem
Hello all,
I try to use a agi script to get a variable from * und put them into a
script which gives me another variablke and put this in *.
My problem is now it seems the var ID is empty coz i always jump into
the result 0 loop.
The $MSN should be in the SetCIDNum.
#!/usr/bin/php -q
<?php
include("/var/lib/asterisk/agi-bin/phpagi.php");
$agi = new AGI();
$ID =
2005 Jul 26
2
sip+oh323 - no voice at sip side
Hello,
I have something like this:
SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN
After calling from SIP to PSTN (and from PSTN to SIP too)
I can't hear anything only in my SIPUSER. At the PSTN side everything is OK.
I have another network with another h323/sip (in the place of asterisk)
and there everything is OK.
In AUDIOCODES logs I see that everything goes
2007 Mar 29
5
SIP RTP Tunnel
Hello,
is it possible to rout ALL RTP Data over Asterisk, like
SIP1 <---RTP---> Asterisk <---RTP---> SIP2
I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;)
Thanx,
Kalle
2011 May 02
7
ATA refuses to answer a call?
I'm kind of at a loss to diagnose problems like this, yet we get them a lot.
- The ATA (Thomson 784 in this particular case) is logged into the
Asterisk server. 'sip show peer' shows their IP address, port, and
useragent.
- The ATA is connected directly to the internet (no NAT, but the sip
configuration has nat=always) and logs in to our server, which is also
directly connected to the
2010 Dec 06
1
[3102] How to rewrite CID name + number?
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.
How can I tell
2005 Aug 18
1
asterisk with odbc
hello
i am trying to use res_odbc for sipuser. my connection
is working. i have checked using isql. even cdr_odbc
is working but i hav problem in res_odbc. i have
created user in sip_buddies table but asterisk is no
getting user from this sip_buddies table.
/etc/asterisk/extconfig.conf
[settings]
sipusers=>odbc,asterisk,sip_buddies
sippeers=>odbc,asterisk,sip_buddies
2007 Jan 17
1
transfer problem
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't work. I have a traditional PBX connected with a zap channel
to Asterisk in the following way:
IAX/SIP client --> Asterisk (FXO) --> (FXS) traditional PBX ---> OFFICE
Phones
Asterisk is connected to the PBX with an internal number configured inside it.
In other words i keep an
2010 Jul 14
2
realtime music on hold
Hello list,
using asterisk 1.4.30.
When setting up the MySQL table 'musiconhold' as described in
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf ,
what is the meaning of the fields :
`*digit*` char(1) NOT NULL default '',
`*sort*` varchar(16) NOT NULL default '',
and what are there default values ?!
What is the default value of :
2005 Jan 13
1
problems with astcc
hello *'s,
Astcc not workin what is correct format for defining
1-database
2-brands
3-trunks
4-routes
i define all these things but not workin may be i define in wrong
format.I have FXO card installed.can anyone implement it and also my
sip phone generates very loud noise wat is that i tried several
settings but not hear any voice just noise.
sip.conf
[general]
context=from-sip
port=5060
2007 Dec 28
1
sip.conf & realtime
Hi -
I'm looking into realtime and I'm having a bit of a problem with the SIP
part.
My review of the posts seems to indicate that I should use realtime static
for the [general] part of my sip.conf including the registration commands:
register=><did>:<secret>@<domain>/<did context>
and use realtime realtime (funny name!) for peers and friends:
[myprovider]
2007 Dec 29
1
Realtime & sip.conf
Hi -
I'm looking into realtime and I'm having a bit of a problem with the SIP part.
My review of the posts seems to indicate that I should use realtime
static for the [general] part of my sip.conf including the
registration commands:
register=><did>:<secret>@<domain>/<did context>
and use realtime realtime (funny name!) for peers and friends:
[myprovider]
2009 Jun 17
1
Incoming Call trouble with new *Now 1.5 setup
Hi All,
I'm having a bit of trouble with my new *NOW setup.
I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from
SimpleSignal.com. Outbound calling works great, but I'm having some trouble
with inbound calls.
First, we would get the "the number you have dialed is not in service" error
on inbound calls. After some googling, I found out that I needed
2006 May 23
3
AGI ?
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a number
the script check to see if it is local if so, go out the ZAP channel. If
the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go
out the SIP channels. Here is a sample of what I have in my script.
#!/usr/bin/perl
use strict;
use
2004 May 25
3
Voice Pulse
Hello:
I am new to the list. I am trying to set up asterisk with voicepulse. I
have a voicepulse username + password, and SIP DID. When I login to
voicepulse, I have this under my devices tab:
Devices
*Login:* Sysxxxxxxx
*Password:* xxxxxxxxxx
*Context:* VPWS
*Connects to:* gw5.voicepulse.com
My question is: Do I need a 2.4.x kernel? Currently I am running
Debian/stable stock 2.2.x ? Has
2012 Aug 14
0
SayUnixTime quandry
Hi Gang,
Hopefully somebody out there has a "doh" for this one. My
dialplan announces the date and time using SayUnixTime. When I run this:
exten => 36225,1,Set(ABA=999999999)
exten => 36225,n,Background(telbank/${ABA}/${CHANNEL(language)}/thetimeis)
exten => 36225,n,sayunixtime(,,Abe 'digits/at' IMP)
I get this CLI output
-- Executing