Title: Open files / socket leak
We're using STABLE CVS-Nv1-0-5-02/24/05 and we've been noticing that sometimes there's a socket leak on REGISTER SIP messages. We've seen it happen only on customers using Sipura SPA2100 ATAs.
If I issue a sip show channels, I see thousands of zombie
Title: cdr_odbc logging insane integer values
I'm having a problem with * (tried both HEAD and STABLE). When logging with cdr_odbc through unixODBC to MySQL, I get insane integer values in the duration, billsec, disposition and amaflags fields. I have enabled MySQL logging, and that's the
Title: Segfault when using res_config_odbc on x86_64
I'm trying to move our asterisk setup from an i686 server to an x86_64 (Dual AMD Opteron) server.
Everything has been manually compiled: MySQL 4.1.10, MyODBC 3.51.11, unixODBC 2.2.10 (because I couldn't find any usable RPMs). And
Title: Asterisk on Solaris 10
Does anyone have experience compiling Asterisk STABLE 1.0.5 on Solaris 10 for x86? I have looked at http://www.voip-info.org/wiki-Asterisk+Solaris+Support but I'm looking for other people's experience in actually using Asterisk under that platform. We only need
Title: HEAD vs STABLE
I am having a somewhat hard time finding out what the current HEAD and STABLE versions of Asterisk are. I'm currently running CVS-HEAD-08/24/04 and I think I need something newer :-)
What are the version numbers of HEAD and STABLE? I was unable to find out by reading
Title: AS5xx0: SS7 and SIP?
We currently use Asterisk to provide a SIP-to-PSTN service. The actual conversion takes place somewhere in a softswitch owned by our SIP-to-PSTN provider, where we have an SS7 link. We would like to do that conversion ourselves.
Is it possible to replace a
Title: Multiple IPs and SIP
This topic has already been brought up lately, but I'd like to inquire if there are any news.
I have 2 IP addresses assigned to my ethernet card (eth0 and eth0:0) that are on 2 different subnets (one public, one private), because our PSTN gateway provider has
Title: R: [Asterisk-Users] problem with zyxel prestige 2002
We also had this problem with the zyxel 2002. Upgrade to the latest firmware, then it will work. Older firmwares had trouble with incoming calls behind NAT.
-Manuel
-Messaggio originale-
Da: Stig Thune [mailto:[EMAIL
Title: Zyxel Prestige 2002/2002L sound quality
Hi everyone,
I've been trying a Zyxel Prestige 2002L ATA with Asterisk, but I have a problem with very bad sound quality (using G711, it sounds very robotic and metallic) and there is a very long delay in the audio. This all doesn't happen with
I have recently posted a message regarding hung SIP channels when using the Monitor()
command. Well, I was wrong.
The channel hanging wasn't caused by the Monitor command. They also hang when they
aren't monitored. The cause seems to be our PSTN gateway provider. When for some
reason their
What is the current stable firmware version?
1.0.5.11
Do the ATA's and the phones use the same firmware?
Yes, it's the same firmware
-Manuel
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
We use the Monitor() command to record all incoming calls to our call center. After
about 100 incoming calls, the Monitor() command starts to hang, as follows:
- a call comes in
- Asterisk starts recording, the -in.wav and -out.wav files are created
- the partys talk
- after a while (between 1
I'm having a problem with some customers sitting behind hopefully SIP aware routers
doing NAT. These routers translate port 5060 to something different (ie. 10001) in
order to be able to connect more than one SIP client on a single NATted LAN.
Unfortunately, after a while the router seems to
I'm having the same problem here. Any solution to this problem?
-Manuel
(sorry for top-posting, I'm having a stupid mail client here)
-Messaggio originale-
Da: Simon Brown [mailto:[EMAIL PROTECTED]
Inviato: giovedì, 1. luglio 2004 02:05
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users]
Is it possible to tell asterisk not to strip the leading 0
of *incoming* MSNs? I use asterisk with i4l and whenever
I get a call from an long-distance party, the leading 0, which
should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the
hi...
here in Italy is almost impossible to set an
invalid cid, if is out of your allowed space.
ie. if you have X numbers on your PRI,
you can only set one of these. nothing more.
on bri you simply cannot do nothing.
just my 2 cents.
In Switzerland CLI is also impossible to spoof -
Does anyone know of a software SIP fax client? Something I can install on a PC which
connects to the asterisk server and sends/receives faxes? Something like XLite - but
to fax instead of to phone.
I know of the fax machine connected to an ATA solution, but that's not really what
I'm looking
I have been playing with res_odbc in these last days, but it doesn't want to work.
This is the output when starting Asterisk, so everything seems OK:
[res_odbc.so] = (ODBC Resource)
== Parsing '/etc/asterisk/res_odbc.conf': Found
Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:132
I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk.
It works ... Partially.
We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA.
First of all, when I make a SIP call to the unit with a simple Dial() command (no r,
so Asterisk
I want to have call forwarding (from the POTS)
turned on at the close of work and turned off
automatically by *.
I would have a look at GotoIfTime:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
That should be much easier than a cron job
Regards
-Manuel
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know
what the parameters mean. I looking for a man page basically.
The problem isn't related to the t in the Dial() command, which enables
Based on th wiki, avoid kernel 2.6 unless you know what you are doing.
Likewise with fedora, which seems to work but needs kernel thread turned off.
Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel
2.4.22-1.2188.nptlsmp on Dual Xeon systems. It has worked
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway
provider supports both, so that's not a problem, and if I force him (the PSTN gateway)
to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider
Try to configure in sip.conf your extensions context like this:
[XXX]
disallow=all
allow=g729
Done that already: but then, the incoming channel (from the user to Asterisk) is
G729, and the outgoing channel (from Asterisk to the PSTN gateway) still remains
ULAW, so Asterisk has to do
Define that per user.
Of course... The user part is not the problem. If I force a user in its extensions to
use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to
the PSTN gateway, doing the transcoding. This is driving me crazy...
-Manuel
When a user calling over the PSTN network calls one of our SIP users with a restricted
number (CLIR), our PSTN gateway is sending us incoming calls with the following
additional headers:
Proxy-Require: privacy
Anonymity: uri
Remote-Party-ID: sip:[EMAIL PROTECTED]:4000;privacy=uri
as opposed
If I understood your initial objective correctly (and I may not have),
the user's phones are negotiating the codec to be used for each rtp session.
Asterisk parameters can be used to dictate rtp sessions between the sip
phone and asterisk, but that won't influence the next step in which the
Did you try having two sip.conf entries for your gateway? Forcing one
with G729 and the other with ulaw? You would obviously need to change
your dialplan accordingly and have each phone configured so that it
would take the proper extension. I have not tried this, it is just
really an
Hmmm, I was thinking about this problem too... What type of gateway are
you using? Is it registering with the Asterisk server? I would try using
two different 'virtual' extensions on the gateway and in sip.conf. That
way you would have full control on how calls from the gw to * are handled.
They don't need to have the same IP. Assign several IP numbers to your
linux box:
ifconfig eth0:1 10.1.1.1 netmask 255.255.255.0
ifconfig eth0:2 10.1.1.2 netmask 255.255.255.0
Sorry guys... These are all great tips, but also this doesn't work: the gateway is not
under my control, it is
Use two separate entries with type=peer and type=user instead of one
entry with type=friend.
Tried that as well. This triggers yet another misbehaviour...
I tried to define 2 peers (for the outgoing calls), one called [gateway-g729] and one
called [gateway-ulaw], each allowing only the codec
If I have ODBC logging enabled (with cdr_odbc), Asterisk logs everything to ODBC *and*
to the CSV file (Master.csv). If I issue a reload, it stops logging to the CSV file,
but continues logging to ODBC.
To have it log to the CSV file again, I have to issue unload cdr_csv.so then load
I have defined a SIP friend without username and secret, only IP-based. I have also
defined an accountcode for that friend, as follows:
[mypeer]
type=friend
host=192.168.0.100
port=5060
context=mycontext
canreinvite=no
accountcode=mypeer
Unfortunately the accountcode for the calls originating
I'm not really being too lucky in the last days. After trying to compile cdr_mysql
with no success, I am switching to cdr_odbc. I have installed unixODBC, iODBC and
MyODBC correctly, I am even able to make queries with isql. But when trying to make
in the cdr directory of the latest CVS, that's
I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get
that error:
*CLI load cdr_odbc.so
Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2:
cannot open shared object file: No such file or directory
Unable to load module
I'm trying to compile cdr_addon_mysql but keep getting this error.
Again, searching the Wiki and the mailing list archive didn't bring up
anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to
switch back to 3.23?
# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Manuel Wenger
Sent: 21 June 2004 12:43
To: [EMAIL PROTECTED]
Subject: R: [Asterisk-Users] Re: cdr_addon_mysql compiling error
I'm trying to compile cdr_addon_mysql but keep getting this error.
Again
By reading the Wiki's I found out that an Asterisk server with many (1)
extensions and/or SIP users can become slow when reloading. But what happens when you
also have many contexts in extensions.conf? More precisely, one context for each SIP
user?
I need this because I will have users
-Messaggio originale-
Da: Kevin Walsh [mailto:[EMAIL PROTECTED]
I don't quite understand your Caller*ID dilemma.
In your sip.conf, you'd have a block for each user, say [abc123].
That's your random username, yes? The same block would also
define the password and other directives.
I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching
the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes,
I'm using MySQL 4.0. Maybe I have to switch back to 3.23?
# make
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql
I am trying to make Asterisk communicate with a voice switch which doesn't need (and
like) authentication on outgoing SIP calls. I have configured it as follows in my
sip.conf:
[myswitch]
type=friend
host=192.168.1.100
port=5060
context=default
canreinvite=no
To dial out using this switch (it
Before starting to look at the problem in Asterisk, make sure that your phone company
has enabled the selective CLIR feature. Otherwise the phone exchange will simply
ignore your request to hide CLIP.
Regards
Manuel
-Messaggio originale-
Da: Pedro Vela [mailto:[EMAIL PROTECTED]
Yes, same here.
-Manuel
-Messaggio originale-
Da: Stephan Wik [mailto:[EMAIL PROTECTED]
Inviato: venerdì, 11. giugno 2004 12:46
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] VoipTalk down?
SIP registration not working, web site down. Anybody else see this?
Stephan
That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was
staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had
choppy audio. After disabling HT, everything was fine again. Nothing measurable,
indeed, but you could definitely hear it. So
]
Oggetto: RE: [Asterisk-Users] Hyperthreading?
What cards was it FXO - cos is it card related this HT problem?
-Original Message-
From: Manuel Wenger [mailto:[EMAIL PROTECTED]
Sent: 01 June 2004 3:18 PM
To: [EMAIL PROTECTED]
Subject: R: [Asterisk-Users] Hyperthreading?
That's the problem we
We are planning to deploy a pretty large asterisk server with many SIP extensions
(might be up to 1 in the future), and I have a few questions:
1) is this possible, or are we running into some kind of limitation in the software
that I wasn't aware of and that I didn't find by browsing
I've just checked out the latest CVS from the 1.0-stable branch, but DateTime() seems
somewhat buggy. It says something like:
Tuesday May 18 11:46 AM 2004
instead of
Tuesday May 18th 2004 at 11:46 AM
(notice the wrong order of the words and the missing th/at)
Did I miss something? Does
Hi Tony,
Try adding fromuser=x, maybe username= isn't enough... Just a guess, it
already solved a few problems for me.
-Manuel
-Messaggio originale-
Da: Tony Hoyle [mailto:[EMAIL PROTECTED]
Inviato: martedì, 18. maggio 2004 13:03
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users]
I've just setup a new asterisk server, and I need to have G.729 working on this
system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc),
but only /dev/sda.
Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really
stupid to replace an entire
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