Hi;
I faced same problem, and I restarted my Routers at
both sides and did not resolve it. Anyway, I am still
inverstigating.
Regards
Bilal
I have a network of offices using Asterisk that are
connected via IAX2
> trunks. The trunks work great for a day or two then
for
Or, you can write your own scripts to generate calls via the Manager
API, or use Asterisk call files (see voip-info.org on this topic).
But, all other things being equal, it is probably preferred to use some
sort of testing framework of the sort mentioned below.
Atis Lezdins wrote:
> On 2/18/08
Naveen Palani wrote:
> Hi,
>
> I have the asterisk-1.4.11 set up installation on my Ubuntu machine.
> When i try making a simple incoming call using xlite softphone. I get
> the following message when i try calling to the number.
>
> *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 proce
Hi,
I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try
making a simple incoming call using xlite softphone. I get the following
message when i try calling to the number.
*CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No
compatible codecs, not ac
On Monday 18 February 2008 19:04:54 John Von Essen wrote:
> Before I attempt to program a system like this, I wanted to see if: A)
> its possible, and B) its too insanely difficult for a perl developer.
>
> My office building has a dialer on the front door so people can call me
> and gain access. T
Thanks for the reply, Lee.
I have one small problem, though... I'm using Trixbox (don't judge me), and
I can't make heads nor tails from its dialplan... I know I should be looking
at extensions*.conf, and something dialing the IAX2 trunk, but I can't find
anything :(
Could you point me in the rig
Greetings,
One of the things that I have been working with off and on is a new event API
for Asterisk. Most of the current infrastructure was written almost a year ago.
However, just recently, I extended this system to be able to allow device states
to be shared within a cluster of Asterisk serve
On Mon, 2008-02-18 at 22:17 +, Ben Willcox wrote:
> We use the automon *1 recording function in asterisk, which allows users
> to record a call if necessary on the fly. Unfortunately there doesn't
> appear to be an easy way for the user to actually access that
> recording.
I've always done thi
On Mon, 2008-02-18 at 23:18 +0100, Alessandro Russo wrote:
> I would like to limit the numbers of inbound h323 connections for
> different extensions, for instance, I've the following rules in my
> dialplan:
>
> exten => 123,1,DIAL(H323/1100)
> exten => 234,1,DIAL(H323/2200)
>
> and I would l
Seysan wrote:
> Hello,
>
> What is the Differences between Asterisk 1.4 and Asterisk 1.6 ?
>
> Also I mean what has made it to be in a separate Branch ?
There are release notes that speak to this.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0
Hello,
What is the Differences between Asterisk 1.4 and Asterisk 1.6 ?
Also I mean what has made it to be in a separate Branch ?
thanks
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I am guessing that 'not yet handled' is not good news.
PaulH
P[ 2] I IND :SETUP oad:383208100 dad:93409098 pid:415 state:none
P[ 2] I SEND:PROCEEDING oad:0383208100 dad:93409098 pid:415
-- Executing Answer("mISDN/2-1", "") in new stack
P[ 2] * ANSWER:
P[ 2] I SEND:CONNECT oad:0383208100 dad
Before I attempt to program a system like this, I wanted to see if: A)
its possible, and B) its too insanely difficult for a perl developer.
My office building has a dialer on the front door so people can call me
and gain access. The dialer on the door has a full keypad, and
basically just ring
sean darcy wrote:
> That is, is port 1 = channel 1 and slot 1?
Yes, they are. However, 'UNCONFIGURED' means you haven't run ztcfg yet,
so Asterisk cannot use the channels.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
_
I'm getting this error on starting:
WARNING[14605]: chan_zap.c:1113 zt_open: Unable to specify channel 1: No
such device or address
[Feb 18 18:19:44] ERROR[14605]: chan_zap.c:8074 mkintf: Unable to open
channel 1: No such device or address
here = 0, tmp->channel = 1, channel = 1
[Feb 18 18:19:44
On Mon, 2008-02-18 at 08:11 -0500, Jared Smith wrote:
> On Mon, 2008-02-18 at 16:35 +1100, Paul Hales wrote:
> > I spoke to a telco tech and he said I had to send a facility
> > codehuh?
> >
> > Anyone with any ideas on this one?
>
> I know there's a setting in zapata.conf called "facilityena
The Asterisk.org development team has released Zaptel versions 1.2.24 and 1.4.9.
Zaptel 1.2.24 Highlights
* Linux kernel 2.6.24 compatibility
* New module parameters for tuning VPMADT032 echo canceller modules
* Improved interrupt handling in the wcte12xp driver
Zaptel 1.4.9 Highlights
* Linu
On 18/02/2008, Tuukka Laurikainen <[EMAIL PROTECTED]> wrote:
>
> Just make sure you have the dsp's necessary installed for the simultaneous
> calls
> on the router you're planning to use.
Thanks every one, having looked at
http://www.cisco.com/warp/public/788/products/1750-vic-issues.html and
h
The Asterisk.org development team has released Asterisk-1.6.0-beta3.
This release contains a number of bug fixes over beta2, as well as a few new
features.
* Added an 'n' option to SpeechBackground to request that the channel not get
answered
* Added a number of new manager actions to improv
Hi to all,
I would like to limit the numbers of inbound h323 connections for different
extensions, for instance, I've the following rules in my dialplan:
exten => 123,1,DIAL(H323/1100)
exten => 234,1,DIAL(H323/2200)
and I would like to limit to 5 the number of h323 connections for exten 123
Hello All,
Our old Lucent Argent system had a feature whereby when you initiate
recording during a call, it would afterwards send the recording as a
voicemail message to the user who initiated the recording.
We use the automon *1 recording function in asterisk, which allows users
to record a call
On Mon, 18 Feb 2008, Stephen Davies wrote:
> I have a network of offices using Asterisk that are connected via IAX2
>> trunks. The trunks work great for a day or two then for no reason at all one
>> end of the trunk will become UNREACHABLE while the other end is still
>> connected. The oving nly w
HI
Thanks for the reply
If you activate debug you will see that you get those warnings because
> Asterisk is trying to check users that only exist in the "sip.conf" file.
>
> OK, but could you be more specific. My sip.conf despite the general config
is completely empty. So what do you mean by "ch
Quoting Tim Johnson <[EMAIL PROTECTED]>:
> I have a SPA3102 which is supposed to be similar. Make sure you leave
> the PSTN --> Subscriber Information --> Display Name blank. Also, in
> your sip.conf file, do not specify any "callerid=" value. ...
It was worth a try, but unfortunately it makes n
On Mon, 2008-02-18 at 19:14 +, Razza wrote:
> Is anyone using a cisco router as an ISDN gateway with Asterisk?
> As you might have seen from a couple of my threads, I have been
> looking at Fritz! and Cologne cards, both of which require
> development against a specific version of asterisk/zapt
I know this is a bit off the thread
But I am trying to see if anyone in here know how to config a AS5300 with 2
T1.
Please contact me off list if you can give me a bit of help
Sam Tam
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I have loaded the SIP firmware for an Avaya 4610sw IP phone and have
successfully registered it to Asterisk BE and Asterisk 1.4.18. I am however
experiencing two issues that I am hoping someone has already overcome.
The first one is that the phone looses its registration from Asterisk every
n
I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware.
I have successfully registered this phone to Asterisk BE as well as Asterisk
1.4.18
Almost everything is working well. Except for two issues.
One of the problems is that the phone looses registration every now and
We use PRI, not BRI, with Cisco gateways and it works great. Rock solid.
Razza wrote:
> Is anyone using a cisco router as an ISDN gateway with Asterisk?
> As you might have seen from a couple of my threads, I have been looking
> at Fritz! and Cologne cards, both of which require development aga
> Is anyone using a cisco router as an ISDN gateway with Asterisk?
> As you might have seen from a couple of my threads, I have been looking at
> Fritz! and Cologne cards, both of which require development against a
> specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive
> and c
There has been some success using the 3810 series routers with a SIP
image as an up to 6 port gateway.
One fellow has even been able to come out of the 3810 T1 port into a
channel bank and derive up to 30 channels, 24 from the T1 and 6 from the
AVM with daughter boards.
No extensive testing has
Are you not sending the exten to iaxmodem like this:
exten => xx,1,dial(IAX2/xxx/${EXTEN},30,r)
I had the same problem, routing wouldnt work, until i passed it the did like
above.
> Message: 3
> Date: Mon, 18 Feb 2008 15:26:31 +0200
> From: Louwrens Benad? <[EMAIL PROTECTED]>
> Subjec
Quoting Jaap Winius <[EMAIL PROTECTED]>:
> Hi list,
>
> Hopefully, some of our Dutch members can help with this one. I'm also
> based in the Netherlands and am using a Sipura (Linksys) SPA-3000
> (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test
> system. It works fine, except
Is anyone using a cisco router as an ISDN gateway with Asterisk?
As you might have seen from a couple of my threads, I have been looking at
Fritz! and Cologne cards, both of which require development against a
specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive
and causes a la
Hi,
Is anyone still using backticks on 1.4? Or is there another way to pull a
variable from a shell script into Asterisk 1.4?
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protocolend does not matter when using MFC/R2, you can totally forget
about it, ignore it etc.
What we need is a trace of the R2 signaling. You can take such by
adding this line to unicall.conf
loglevel=255
Then make sure you have enabled max debugging in logger.conf in
Asterisk. When the failur
Is it possible in Asterisk 1.4 to log by somehow the estimated roundtrip
time (RTT) between server and some peer, which Asterisk computes based on
the sending of OPTIONS and the receiving of the responses to those OPTIONS?
Regards,
Ricardo Carvalho.
___
Hello,
If you activate debug you will see that you get those warnings because
Asterisk is trying to check users that only exist in the "sip.conf" file.
PLL.
Original Message
Subject: [asterisk-users] Asterisk reltime mode with Postgresql
From: Andrew Nowrot <[EMAIL PROTECTED]>
Louwrens Benadé wrote:
> NDID=s
>
> What the hell!? Why is 'NDID=s'?
Probably because at the point when your dialplan sends the call to
iaxmodem ${EXTEN} is "s".
Thanks,
Lee.
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Hi list
I'm havining problems with a E1 Digium Card (TE120) here are the
description of the problem:
Case 1: Zaptel 1.4.8 Kernel 2.6.22
The system start working correctly, but aleatory the asterisk
prosess give a kernel panic with the messages:
Process ast
Hello,
I'm experimenting with Asterisk and MySQL.
Up to know I've just put iax.conf in a MySQL database and it seems to
work: when a Iax2 client registers the corrispondent row in db is
updated. Good.
However when I have many asterisk boxes pointing to the same db a
problem arises: I need an additi
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Kevin P. Fleming
> Sent: Monday, February 18, 2008 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] is encrypted iax safe and secure?
>
> Steve J
On Sat, Feb 16, 2008 at 12:07:39PM -0500, sean darcy wrote:
> Building zaptel-1.4 from today's svn against kernel 2.6.25-git2:
What version is that, exactly?
zaptel includes a script to build vs. any specific tag from Linus' git
repo. What tag is it?
>
>
> Building modules, stage 2
sean darcy wrote:
> Should I worry? If so, what do I do?
We don't normally support Zaptel building against unreleased kernels,
because they have a habit of changing before getting to release state,
and in the past this has caused us to have to change Zaptel more than
once during a kernel release
Thomas Kenyon wrote:
> On a very similar note, I have a server that only has PCI-64bit 33MHz
> slots left (don't know if 5V or 3.3V).
>
> Can I place a TDM400P in there? (please no response telling me that
> there are better cards, I'm sure there are but there aren't any better
> card that I a
Steve Johnson wrote:
> Of course *it would be nice if* the IAX2 authentication parameters
> were also encrypted, so that there was no danger of a 3rd party
> hijacking your connection and generating a bunch of extra charges.
Can you elaborate? I don't see any way that a connection can be
'hijacked
Hi list,
Hopefully, some of our Dutch members can help with this one. I'm also
based in the Netherlands and am using a Sipura (Linksys) SPA-3000
(firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test
system. It works fine, except that the Called ID (CID) is not working.
I'm
Hi Jakub
could you please post the zaptel.conf and asterisk cli unicall channel error
plus what is version of unicall and zaptel are u install and i think you miss
the "protocolend= option (cpe or net??)" line in the uncall.conf
ayman
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Mon, 1
Hi list,
The default automon (touch monitor) output file name format is:
auto-epoch-caller-callee.wav
A variable is available to modify the second half:
auto-epoch-${TOUCH_MONITOR}.wav
But, I can't modify the first half, 'auto-epoch-', with any variables
that I know of, including ${M
I have a network of offices using Asterisk that are connected via IAX2
> trunks. The trunks work great for a day or two then for no reason at all one
> end of the trunk will become UNREACHABLE while the other end is still
> connected. The oving nly way to fix the problem is to shutdown Asterisk
> c
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I would try make clean/make/make install.
also add tor2 to the black list and remove it from any zaptel init stuff.
Finally once your systems up (note asterisk wont be) try loading the
module with insmod.
If it panics this may give you a better opur
Pardon my slight outburst. I've just been struggling for a very long time to
get fax going on this system.
I'm running Trixbox 2.4.2, Asterisk 1.4.17, with iaxmodem 0.3.2-14 and
hylafax 20080108:4.4.4-1rhel5.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf O
Hi guys
I have a lovely little irritation on my hands. Ive been trying to set up
DID-based routing in Hylafax, and Ive come across one hell of a headache:
>From minicom:
AT+VRID=1
OK
DATE=0218
TIME=1524
NAME=xx
NMBR=xx
ANID=
USER=
PASS=
CDID=
NDID=s
What the hell!? Wh
On Mon, 2008-02-18 at 16:35 +1100, Paul Hales wrote:
> I spoke to a telco tech and he said I had to send a facility
> codehuh?
>
> Anyone with any ideas on this one?
I know there's a setting in zapata.conf called "facilityenable" -- have
you tried setting it to a value of "yes"? I'm not sure
Re: Contents of asterisk-users submissions
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On 18/02/2008, srinivas Antarvedi <[EMAIL PROTECTED]> wrote:
>
> Hello all,
>
> I am struggling with sending voicemail as an attachement in Email.
>
> When i have given the email like [EMAIL PROTECTED] it is delivering
> to my gamil account perfectly(of course to spam folder).
>
> But when i given
Old behaviour doesn't end current Cdr
-> At the end of the call, you have 2 Cdrs, 1 before et 1 after
ForkCdr call. Each CDR keep trace of the duration of the Call.
I want this behaviour in order to generate 2 CDRs, 1 for my customer,
and 1 on my reseller
Set(CDR(accountcode)=mycusto
On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I want to have a PC-based real-time VoIP bulk call generator (including both
> SIP signaling and RTP generation)
>
> for stress testing and precise analysis of the VoIP network equipment.
>
>
>
> Do any one knows a free program can do t
On 2/18/08, thieums <[EMAIL PROTECTED]> wrote:
> Hello,
> I'm looking for a way to restore old behaviour (before Arkadia patch
> #0010668) of ForkCDR application in 1.4.18
> I've done some research directly in the code (cdr.c & forkcdr.c), but
> can't find any flag.
> I am just f*c*ed or do you
On Feb 17, 2008 11:30 PM, srinivas Antarvedi
<[EMAIL PROTECTED]> wrote:
> Hello all,
>
> I am struggling with sending voicemail as an attachement in Email.
>
> When i have given the email like [EMAIL PROTECTED] it is delivering
> to my gamil account perfectly(of course to spam folder).
>
> But when
Hi all,
I have configured my asterisk server as gateway and gatekeeper both.
I am trying to call using SIP agent to h.323 agent but it is not successful.
I have configured ooh323.conf as
gateway=yes
gatekeeper=10.17.112.12
Still its not working.
What configuration file I need to change for
In article <[EMAIL PROTECTED]>,
<[EMAIL PROTECTED]> wrote:
> hi.
>
> could somebody explain how exactly the following parameters
> in zapata.conf work:
>
> pridialplan
> prilocaldialplan
> internationalprefix
> nationalprefix
> localprefix
> privateprefix
> unknownprefix
They are all to do with
I want to have a PC-based real-time VoIP bulk call generator (including both
SIP signaling and RTP generation)
for stress testing and precise analysis of the VoIP network equipment.
Do any one knows a free program can do that .
Regards
***
Hello,
I'm looking for a way to restore old behaviour (before Arkadia patch
#0010668) of ForkCDR application in 1.4.18
I've done some research directly in the code (cdr.c & forkcdr.c), but
can't find any flag.
I am just f*c*ed or do you have something to suggest ? :)
Thank you for help.
Math
HI all,
How can I modify the from address in sip message? Say, I will a sip
account 1234. I want to change the from address in sip message of
this sip account to 4321.
From: "4321" ;tag=as5b42e6
ango
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The Vancouver Linux User Group is holding a "Virtualization Round Table"
Monday (Feb 18) evening at the BC Institute of Technology discussing
some of the different approaches to server virtualization. I'll be
speaking about using OpenVZ to provide virtual servers used to host
multiple instances
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