Displaying 20 results from an estimated 2000 matches similar to: "NAT issue (i think?)"
2009 Sep 18
3
DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729
2015 Feb 16
1
Asterisk 11.6. SIP realtime lost peers after 'sip reload'
Hi, list.
We have a problem with loss peers after 'sip reload', our configuration:
Asterisk 11.6-cert1, SIP realtime peers, sip.conf:
- rtcachefriends=yes
- rtsavesysname=yes
- rtupdate=yes
- rtautoclear=yes
When we do 'sip reload' , peers are removing from available.
Before `sip reload` :
srv-pbx2*CLI> sip show peers
Name/username Host
2010 Apr 17
1
Realtime changes not reflected realtime
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<head>
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<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
Using Asterisk 1.4.25.1<br>
Using realtime sip_buddies<br>
<br>
I notice
2010 Jun 04
1
originating a sip call from the CLI
Hello again!
I just got a SIP account and it seems - from a config on the net -, that
I've configured it correctly. But I get no call to the outside. Registration
was OK.
I tried:
channel originate sip/1/echo at iptel.org Application ...
I see the channel active for a while, but no call gets established.
In my config I have defined the section [iptel] for the outgoing call and I
2011 Jan 02
1
Realtime SIP, multiple AX servers question
We have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way.
Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as "regseconds", "lastms", "ipadr", etc. However, with
2008 Mar 13
1
sip.conf help, inbound calls fall to last specified context
First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA?
Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to
2007 Mar 02
4
rtsavesysname not working in 1.4
I am trying to have asterisk update the system name in my realtime
peers, but it does not seem to be working. Here is what I've done so
far.
- added systemname => mysystemname in asterisk.conf
- set rtsavesysname=yes in sip.conf.
- created a table called "sysname" in my peers table in mysql
- restarted asterisk
- rebooted my phone to force a re-register
Is there something
2009 Aug 25
0
DTMF duplicated when Waitexten
Hello,
I have a problem of DTMF duplication.
I receive call from my provider with SIP protocol. These calls pass
through an interactive voice menu, using the application Waitexten to
enter a client code. The menu works fine, but sometimes I have DTMF
duplication that prevent proper code entry. All DTMF come twice.
my sip.conf
-----------
[general]
context=default
allowguest=no
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2008 Nov 05
0
SIP Qualify is not working with Postgres
Hello.
I'm using Asterisk 1.4.22 with Postgres 8.3 in a Ubuntu 8.04 Server.
I configured Asterisk to get sip from Postgres, and set qualify for all sips
as yes, but the sip show peers command show the status of the peers as
UNKNOWN
srvcentral*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
Realtime
4900/4900 (Unspecified) D
2013 Feb 17
0
Can Cisco 5XX phones share asterisk phone directory?
Hi!
Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone
directories, and if so, how?
Thanks in advance!
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2006 Apr 12
1
Where is the difference sip.conf - Real-time ?
I have two phones (111 and 112) on a LAN, and I have on a users site a
phone 333.
phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are identically set-up (just user name = phone number and
password are different)
sip.conf (for 111 - all
2010 Jun 29
1
Can't call my extension
Hi,
I managed to get a remote extension to work through a router which can now
call all the other local extensions in asterisk. For some reason, nobody
can call me back. They get failed upon trying. Keep thinking there must be
some caller group to which I need be added. Or perhaps I need to add the IP
address of this phone to the sip.conf file? Please let me know. Thanks.
Nick
2010 Jul 14
2
BLF with Realtime
Hello Asterisk community,
I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?
I'va my dialplan with Realtime
Thanks in advance
--
Saludos
Danny Dias
SkypeID: danny.dias1
2007 Sep 06
7
SIP Debugging to separate log file
Hello, I'm working with our SIP provider to nail down some call quality issues
we're having, and they've asked me to provide SIP debug log files from our
asterisk server. Is there a way to make asterisk 1.4 output only SIP
debugging to a specific log file? Or it is best just to use tcpdump?
Thank you!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY
2007 Feb 14
6
Fax with T.38
Hi all,
I install the last version of Asterisk and I tried to send faxes, but
nothing works.
Here is my configuration:
Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA
<----> Analog Fax 2
I tried Analog Fax 2 -> Analog Fax but nothing works!!
In the Patton configuration I put G711 and no silence suppression.
In asterisk I have
2006 Jan 18
0
rtcachefriends and REALTIME + MWI
Hi,
Is there something wrong with REALTIME (ARA) when used with
rtcachefriends parameter?
In my sip.conf (Asterisk 1.2.0):
rtcachefriends=yes
rtupdate=yes
rtautoclear=yes
Desired configuration is realtime configuration (via odbc) for SIP
phones + MWI. Realtime means the following: when I make changes to db
they should apply with no extra commands executed in CLI.
In order to use MWI with
2013 Jun 02
1
Asterisk T.38 Pass-Through doesn't work
What I have is:
* Asterisk 1.8.10.1~dfsg-1ubuntu1,
* SPA112 ATA with analog fax in 1-st FXS port connected,
* SIP trunk with provider supporting T.38.
My network looks like this:
* spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in
neighbouring LANs,
* Asterisk connects to the provider (80.75.130.136) via router
(82.200.7.184). Router has full DNAT to Asterisk server.
What happens?
2005 Aug 16
6
realtime caching
Can anyone shed some light on realtime caching?
My desired behavior is that MWI works with realtime
voicemail/sip/extensions AND updates to the database take place on the
next call to the extensions.
Right now I have rtcachefriends=yes, and MWI works, but updates to the
database for a cached user seem to still require a reload.
It is my understating that removing rtcachefriends will
2007 Oct 09
2
Asterisk Realtime woes
I have configured asterisk realtime to work with two servers and a seperate MySQL DB.
Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are
1. On the same network
2. Behind a NAT and connected to the same asterisk server as the caller.
However I need this configuration to work for "NAT-ed" clients on different asterisk