The company that looks after my clients internal phone system has a
problem with logging in to the PABX using their data modem.
Connection looks like this
ISDN PRA from telco - Asterisk - SIP Trunk to my clients Asterisk -
my clients Asterisk - E1 port to his old PABX
I am planning to use
Hi
I am trying to get faxes to work with Asterisk and ReceiveFax. The problem
is that I am getting only about 60% of success (for example two of six fax
transmissions always fails). The most common errors are Error : Unexpected
DCN after requested retransmission and Error : Disconnected after
Hi
I am using asterisk 10.1.3
When user hangs up during playing audio files in Background used in
realtime extensions with postgres I am getting this log on the
console:
[Feb 1 23:04:54] ERROR[4831]: res_config_pgsql.c:166 _pgsql_exec:
PostgreSQL RealTime: Failed to query
On 1 February 2012 21:26, Andrzej Nowrot anow...@interia.pl wrote:
Hi
I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime:
[somecontext]
exten = someexten1..
exten = someexten2..
exten = someexten3..
exten = someexten4..
Hi
I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime:
[somecontext]
exten = someexten1..
exten = someexten2..
exten = someexten3..
exten = someexten4..
switch = Realtime/${CONTEXT}@extensions
switch statement was executed after
Hi
I am using Asterisk-1.6.1.6. Recently I had converted billsec and
duration fields in my postgres 8.4 database from integer to numeric. I
wanted to have better accuracy in calculating duration of the call.
When the call is answered I can see better precision (for example
298.758421 sec), The
Hi
I know that this topic was on the list maybe dozen of times. But I
have a question regarding the fax support in asterisk, because all the
information I could get does not give me the clear view of if. I read
that Asterisk 1.8 will have strong fax (t.38) support, but I want to
know if these
Hi
I am having a problem with extension matching in asterisk (I am using
asterisk 1.6.0.6). Is there a difference between extensions matching
in realtime architecture and extensions matching in extensions.conf
file.
For example when I have these two extensions:
-- _0699[134]X
--
Hi
Many thanks for your help.
I managed to solve the problem by rewriting the dialplan. I have split
the line _06[069] into three different extensions 060, 066 and
069 and now asterisk is matching the numbers as I expect it to do.
Maybe it is not the clean solution, but ;).
Many
Hi
OK I have upgraded to 1.6.0.9 and it looks like the issue disappeared,
but I am facing another problem yesterday my asterisk gave me this
into the logs:
[May 6 17:25:53] ERROR[20625] channel.c: ast_read() called with no
recorded file descriptor.
Message was repeated x times and cause my
Hi
I am using asterisk-1.6.0.6 and I have noticed strange behaviour
lately. When a user ends his call asterisk executes twice the h
extensions (in my case this is the AGI script) and writes this to the
logs:
cdr.c: CDR on channel 'SIP/xx-b6623038' already posted.
and after that it crashes
OK I will do that. i let you know about the results.
Cheers
On Wed, Apr 29, 2009 at 9:21 PM, Barry L. Kline blkl...@attglobal.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Andrew Nowrot wrote:
This had happened twice so far. Does anyone know what is causing this.?
Start
Hi
Looks like it was it. Now it works OK. Thanks for help
Cheers
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Hi
I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial
out from manager's console and with Asterisk 1.4.X this settings were OK.
Action: Originate
Channel: SIP/384
Context: main
Exten: 102
Priority: 1
Callerid: 384
I could dial out, but with asterisk 1.6 I get this error.
Hi
Thanks for so fast reply, but I already have this part like this:
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, Channel);
int timeout = atoi(astman_get_header(m, Timeout));
I did not need to change the code. My manager.c already has all the lines
you specified that are wrong.
did you re compile and re installed?
make
make install
after the code change?
david
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Hi
I have a problem upgrading to asterisk version higher than 1.4.18. Yesterday
I tried to install the 1.4.21.2 and everything was OK for 3 hours since the
upgrade, but after that time asterisk started generating 100% load. The same
happened with 1.4.20.X and with 1.4.19.X. The biggest problem is
Hi
I have a problem with dtmf recognition an analog lines connected to Sangoma
A200. The digits (in most cases the first one) are doubled and so my IVR is
useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but
nothing worked. I also noticed one thing it only happens during
Hi
I am having a problem with proper registration to asterisk through IAX. The
peer (which is Iaxmodem) suppose to register to the server each 60 sec and
it is doing so, but the server is aware of the registration only for first
ten seconds and after that time ipaddr and regsec fields in database
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 check
Hi
I am having problems with Asterisk 1.4.18 and realtime architecture. I use
Postgresql-8.3 as the database.
Everything works OK; all sip phones (their configs are in the database) are
able to register to the server and I can make calls (dialplan is in the
database), but each time Asterisk reads
Hi
Thanks for reply
Yes, there's a change. For me it's completely unacceptable, so i
reverted the patch (http://bugs.digium.com/view.php?id=10659).
For me too. This bug occur in September. Is it still present in asterisk
1.4.12.1. I also have asterisk 1.4.4 on a different box and there
Hi
I have a question if there was a major change in CDR?
Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre
happened. After the upgrade I have no call details in the cdr table when the
call did not go through because of for example: Unable to create the channel
of type Sip -
Hi
I am trying to build reliable fax solution with asterisk, iaxmodem and
hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3
1.2GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After
installing
the newest zaptel and wanpipe-3.1.0 beta I did zttest and it didn't give
On 7/7/07, Lee Howard [EMAIL PROTECTED] wrote:
HI
Are you having trouble with fax? Rumor is it that the Sangoma hardware
isn't as needy this way as is the Diguim. I'm not sure about that,
though.
I heard that too, but unfortunately I have some problems with incoming
faxes (but only when
Hi
The kernel timer shouldn't be relevant. The timing should come from the
card, and not from ztdummy. Make sure that the timing comes from the
card and not from ztdummy.
I don't load the ztdummy module, so timing is taken only from the card. The
only reason I put rtc in my kernel is that I
Hello,
I am trying to install bristuff-0.3.0-PRE-1x.tar.gz on debian with kernel
2.6.19.2 and I've got some errors connected with XPP. I was wondering
if somebody managed to install bristuff with this kernel or any kind
of kernel 2.6.19. The bristuff mentioned above contains zaptel 1.2.10 not
, Jan 18, 2007 at 01:56:22PM +0200, Tzafrir Cohen wrote:
On Thu, Jan 18, 2007 at 11:46:01AM +0100, Andrew Nowrot wrote:
Hello,
I am trying to install bristuff-0.3.0-PRE-1x.tar.gz on debian with
kernel
2.6.19.2 and I've got some errors connected with XPP. I was wondering
if somebody managed
Hi Has anyone used the AudioCodes MP-20x?I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive.
RegardsAndrew
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HiTry inkeys instead of inkey and you should be fine.Regards Andrew
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HiI change the kernel to 2.6.14.7, but unfortunately the problem still exist.The
messages empty HDLC frame or bad CRC received appear only when there
is not traffic on card (0 active calls). It never happens during a
call. Strange?!?
Any other tips are gladly expected :).CheersAndrew
On 8/25/06, Stelios Koroneos [EMAIL PROTECTED] wrote:
Its
working for me with no errors.
*
1.2.10 bristuff 0.3.0-pre1s with kernel 2.6.15.4.
My
setup is kind of special as its build with Openembedded and runs from a CF on
a [EMAIL PROTECTED]
Recently i was able to port *+bristuff +
or are you bound to bristuff because you need speciall features of this?Well you are right. Bristuff has more features than mISDN.After loading the florz patch messages in kernlog turn into this
Aug 24 10:18:08 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:20:45
HiCan anyone confirm a working asterisk 1.2 from bristuff with 1 port PCI, hfc-s based ISDN card (zaphfc driver). If so, could you send your configuration. I mean OS (linux distribution) type, kernel version.
Thanks in advanceCheersAndrew
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HiI am trying to set up * box with the ISDN hfc-s cards. One in NT mode and two in TE. I am using thebristuff-0.3.0-PRE-1r.tar.gz
. The installation went well, but soon after the zaphfc was loaded I started to receive these message in kernlog:Aug 23 21:00:08 asterisk kernel: zaphfc: empty HDLC
HiThanks for your reply.I will check it first thing in a morning and of course will let you know about results.Cheers
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HiThe newest bristuff didn't change anything. Still the same. I was wondering if this is happening only to me or not. Does anyone has the same problem? Maybe I am messing something when loading the modules.
Does anyone have any other tips.Andrew
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Hi I am sending the results of my research to the list. Unfortunately any combination of hangupcause worked :(.But I also try this L option on other machine, on one of my Zap channels and this time L worked perfectly. The channel went to hangup state and Asterisk executed the DeadAGI.
So I guess
HiI have a problem with Dial application. The dialplan looks like this:;exten = x,1,Dial(Sip/|30|L(6:3:1))exten = x,2,Hangup()exten = h,1,DadAGI()
;The call is limited to 60 sec and after that time the conversation stops, but Asterisk never reach the h extension.I
What does the CLI show when you make the call? That might help in diagnosingyour problem.
FlynnHi Flynn The situation looks like this:exten = _0800X.,1,AGI(/usr/share/asterisk/agi-bin/checklimit.php|${CALLERIDNUM}|${CONTEXT})exten = _0800X.,2,GotoIf($[${code} = 0
Thanks for all repliesI noticed that L option does not hangup the call it only limits the call. (In my case the h extension isn't executed). S option can do that (Asterisk reach the h extension)L(x:y:z) - do not hang up the call after x sec.
S(x) - hangup the call after x sec.I also noticed that
On 6/27/06, William Piper [EMAIL PROTECTED] wrote:
Although I've never tried it along withthe L option, you couldtry absolutetimeout:
exten = x,1,AbsoluteTimeout(6)
exten = x,2,Dial(Sip/|30|L(6:3:1))I didn't help still the same :(.
On 6/27/06, William Piper [EMAIL PROTECTED] wrote:
Well, It was worth a shot.
Perhaps doing a some variation of the HANGUPCAUSE variable.
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause
exten = x,2,Dial(Sip/|30|gL(6:3:1))
exten =
Hi,I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box:-- motherboard Intel E7210 + Hence Rapids-- processor P4 3.0 GHz-- RAM 2x512 MB DDR ECC-- network interface Intel 82541 GI
Is this configuration enough to handle 30 users at the same time. I am
Thanks for all replies Now, if you system will be accessible both
from inside (LAN) and outside (Internet), I would advice a second network card (10/100)Actually the machine has two interfaces - 1000 and 100 Mbit/s
CheersAndrew
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Hi,I am still struggling with the E1 cardDoes anyone has some experience with sangoma E1 card? I have this card in soekris net 4801. First I was runnig it with deactivated DMA and I was receiving overruns (even with no channels in use). Then I enabled the DMA. Now I have the overruns only
Hi,I finally found an ATA which works really well with asterisk and its application alarmreceiver. Frankly it works just like the TDM card. It is Soundwin S800 series ATA.CheersAndrew
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Hi,Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box.Cheers Andrew
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Hi,Thanks for all repliesCan anyone tell me if it is possible to send the SMS through
this card directly from extensions.conf with some application that
takes the text string and converts it to SMS and which colaborates with junghanns card.
CheersAndrew
Hi, I will try that thanks.Andrew
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Hi,I am trying to integrate Asterisk with traditional phone central, this issue is sometimes tough. After some testing and measuring I think what is bothering my Asterisk; I need to dial a number digit after digit and not the whole string, so for example:
1, 2, 3, 4, 5, 6and not:123456How can I do
HiI sent this earlier, but it was late and I haven't saw any reply. Maybe now I will have more luckDoes anyone know the correct settings of zapata.conf
and zaptel.conf that are needed to connect two asterisk boxes over E1. I am trying to (just for testing purposes) connect two * ( A and B ) boxes
HiDoes anyone know the correct settings of zapata.conf and zaptel.conf that are needed to connect two asterisk boxes over E1. I am trying to (just for testing purposes) connect two * ( A and B ) boxes over E1 link and IAX as well. Both are Soekris 4801 and have Sangma A101U cards. The situation
Hi,I need to have the information about the current IP address of the user. I want to know IP address from which user is registered to Asterisk server. Is it possible with Asterisk to log this information to the database or file? Does anyone can give me some info about this issue? Thanks in
Hi,Thanks for so fast replyOk I know about this but actually I am thinking about logging the IP address of a user in realtime. Each time the user changes his location and register Asterisk will log the time and IP address. Is it possible?
Best wishes
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Hi,I already try that, unfortunately with no success. I wonder what is wrong?Cheers
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Hi,I am trying to connect Asterisk to traditional central. It must be done over ISDN. I am utilizing Asterisk 1.0.9 from bristuff RC8p. I was able to communicate them with each other, but I have a problem with callerid. The traditional central does not recognize the callerid from the phones
Hi,Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked).
Sometimes alarmreceiver is able to get some
Hi,Thanks for so fast reply.Now, rather than just being a nay-sayer, let me refer you to the BoschC900V2 device. It takes a signal from just about any panel and converts
it into IP to be received by a Bosch receiver.Is it possible to connect C900V2 with Asterisk, (did you do such a thing, did you
Hi,Like you said, local connections work OK. Actually I find the problem , it was something I exclude at the beginning - the bandwidth. Some wiseguy created a 80 kbit/s upload queue.But the ISDN could also cause this problem you never know.
Sorry to bother you.
Hi,I have small issue with Asterisk. My customers complaining that sometimes (not always) the outgoing voice (the voice which can be heard by the user a the other end) quality is very low (stutter and sudden clicks). The problem exist in only-IP configuration and in IPtoTDM connections as well. I
Hi,Thanks foe so fast replyI have old Asterisk 1.0.7 which is running on Intel Pentium 4 3GHz. The average load is less then 30% but this voice problem is happening even with one active call.
The other machine with Asterisk 1.2.4 and zaptel 1.2.3 (the same CPU and load) sometimes behave in the
Hi,I use Asterisk with junghanns.net bristuff. My PSTN technology is ISDN and I use the zaphfc ISDN card in TE mode so it synchronize itself to the clock of connected ISDN line.
CheersAndrew
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Hi,Yeah, I think it was all about thew zap channelsBut what opportunities I have when I need to connect two or more Asterisk boxes. IAX, SIP or what?What is most efficient.CheersAndrew
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Hi, Where are you pulling this number from? (other than the obvious traditional 2^8)?
That is not my imagination ;).Actually I talked with a guy who was one of the designers of Asterisk. He told me about this limitation but I don't know if he was talking about Zap channels only or in general. I
Hi, It does sound like a typical case of urban legend, where Zap is limited
to 256 channels becomes Asterisk is limited to 256 channels. Asterisk!= Zap.I've never said that Asterisk is limited to 256 channels. I only asked a question. That is the main reason of this list isn't it?
But leave the
HiIn my environment I have to connect 6 * boxes with each other so IAX is probably the best solutionThanksCheers
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Hi,Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box?I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. What can I do when I need to have more calls than that. I read about connecting Asterisk boxes
Hi,I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this:
--NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
--NOTICE [1197]: PRI
Hi,
My config files look like this:
zapata.conf:
switchtype = euroisdnsignalling = bri_cpe_ptmppridialplan = localechocancel = yesimmediate = nousecallerid = yes
group = 1context
Hi,I certainly don't want to integrate fax-e-mail support into spandsp.I think our problem is not connected with spandsp and fax - email integration. All the applications I mean spandsp txfax and rxfax are enough to have emial - fax functionality in Asterisk. I wrote a program which allows me to
Hi,I've been struggling with this for a quite long time. Maybe I am not the first asterisk user with this problem, (I try to search on google, but I didn't find anything good). My point is:I try to set up * to work as a fax server. Each incoming fax (from PSTN) should be received on email. Luckily
Hi,Few days ago I installed Email2fax application on my Asterisk box. I works but not in 100 %. Sometimes (to be certain quite often) I don't receive the whole fax. My fax machine cuts off transmission in 1/2 or 1/3 of the page. I read about it on a wiki and some user lists and people say that
Hi,The situation looks like this:Email with attached .pdf file --- Asterisk 1.2 box with qmail, connected to PSTN by ISDN HFC-S card, with a_law codec--- PSTN (ISDN)--- Another Asterisk with the same card and codec --- fax machine connected to Asterisk by Digium card.
Thanks for so fast
I have another issue I want mention about.Email2fax application can send you a confirmation email; suppose that you are sending the email, it goes to remote fax machine and you are receiving the confirmation email; but the remote fax machine is out of paper. In traditional situation (when you
Hi,I'm trying to set up asterisk 1,2 with Festival and everything works fine until I install additional languages. When I dial appropriate extension I get something like this:Jan 2 10:43:06 WARNING[836]: app_festival.c:484 festival_exec: Festival returned LP : cstr_pl_em_diphone
Does anyone know
Hi,
I finally managed to install * on Via. It seems that everything works.
Thanks for help.
Cheers
Andrew
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Hi,
Does anyone has some experience in installing * on Via Epia. I am
struggling with it for about two days. And when I finally managed to
install asterisk 1.0.9 after starting it I get this error or whatever:
- Illegal instruction
I changed the variable in makefile to i586 (I also tried
Hi,
Could you specify the amount of makefiles because I use * from
Bristuff and only changed the makefile in asterisk directory. What
others makefiles should I change?
Andrew
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Hi,
I use VIA-C3 Processor family for ezra CPU. Does it make my situation
any better? I managed to compile a new kernel 2.4.30 on this Via Epia.
I have also installed Asterisk with no problems but the after the
start I get -- illegal instruction8(.
If the PROC=i5(6)86 will not change
Hi,
Exactly I want to switch landscape to portrait. Where can I switch it
and what (bash script) can I use for it?
Cheers
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More specifically, you can make it work using an ATA or a TDM400P
card with an fxs port, but it is not likely to be reliable. If you
send a few faxes here and there, that shouldn't be a big deal. If you
are talking about an office where lots of faxing is done, the lack of
reliability will
Hi,
Does anyone know if something like H standard extension exists in
Asterisk (I'm not talking about h standard extension). If yes what
does it do and what is the difference in comparison to h standard
extension.
One more thing; when I put h extension in my dial plan:
exten =
Hi,
Does anyone know if there is a new version of Bristuff for Asterisk 1.2 stable.
If yes, where can I find it and of course download it :).
Cheers
Andrew
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Hi,
I'm trying to translate the voicemail application to my local
language. I want to translate the notification email which Asterisk
send when you have new massages. Where I can find this file ??
Cheers to all
Andrew
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Hi,
I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:
[general]
parkext = 100
parkpos = 1-5
context = parkedcalls
parkingtime = 100
transferdigittimeout = 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark =
Hi,
Thank for the Email
I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?
Best wishes
Andrew
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Hi,
I was wondering if I could use Asterisk logo in my PBX system which I
want to introduce in my local market. Does anyone know if I must fill
some legal issues which let me use this logo.
Best regards
Andrew
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Hi,
I try to set up planet VIP-050 with asterisk. Everything works fine
instead of the call transfer. When I press # console says something
like this:
Oct 5 11:11:20 DEBUG[25104]: chan_sip.c: sip_rtp_read: Oooh,
format changed to 1024
Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144
Hi,
Can anyone tell me if Asterisk sets some variable when doing a call
parking (when someone presses an exten set in features.conf). In can't
find this information on a wiki.
Cheers ;)
Andrew
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Hi,
In my dialplan I'm using a WaitExten() command. It works only with Zap
phones. When I dial this command with Sip phone asterisk do nothing.
Should I put extra definition in sip.conf to make this work with Sip
phones?
Thanks in advance
Cheers
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Hi,
I'm quite new to all asterisk issues. Unfortunately I already have
some problems.
I use ISDN phone and my colleague has got a normal phone connected to
Linksys PAP-2 VoIP Gateway. These two phones are in the same call and
pickup group. With the phone connected to Linksys I'm able to pickup
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