I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
,the following error appears
From where I can get the missing rpms .or kernel source
grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file
or directory
grep:
Trying to launch my first dial out to Teliax and getting this error
[May 29 03:08:06] WARNING[1955]: pbx.c:4644 add_pri: Unable to register
extension '204', priority 2 in 'brad', already in use
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp
You need to install linux kernel headers
2007/5/29, Khaled Chehab [EMAIL PROTECTED]:
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2,the
following error appears
From where I can get the missing rpms .or kernel source
grep:
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
,the following error appears
From where I can get the missing rpms .or kernel source ,or the kernel
header
Generating input for menuselect ...
grep: /include/linux/autoconf.h: No such file or directory
make[1]: Entering
http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS
http://www.voip-info.org/wiki/view/Asterisk+CentOS-4.0+Zaptel
2007/5/29, Khaled Chehab [EMAIL PROTECTED]:
I am using centos 4.4 server cd ,when I am trying to compile zaptel
1.4.2 ,the following error appears
From where I
I am using centos 4.4 ,when I am compiling zapltel using l make linux26
,error accrued ,what s missing
[EMAIL PROTECTED] zaptel]# make linux26
grep: /include/linux/autoconf.h: No such file or directory
make: *** No rule to make target `linux26'. Stop.
Regards
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
I have comiled stats on my asterisk pbx over the last three years.
All spam faxes arrive perfectly readable. For
Hello everybody, I am 20 days with the same item and I can't configure it.
I want to know if someone has the Billion ISDN card on a Debian Etch,
because everybody tells me to do that, then the other one but no one has the
same configuration.
If some one has the same configuration (Billion +
randulo wrote:
All spam faxes arrive perfectly readable. For actual documents faxed
by customers, one in 5 work. Because of this I removed the PDF
Man that's awful!
My experience is just the opposite. I might have a bad fax2pdf
conversion once a month. I review the pdfs being archived on
1. Read the error No such file or...
Do a yum install autoconf
2. And the other error No rule to make target `linux26'
just use make NOT make linux26 since it seems that that is removed!
3. Get rid of that disclaimer !
This will
Hi
The various bits of instruction out there on compiling Zaptel on 2.6 seem to
be a bit misleading.
With the latest versions there is no need to run make linux26
Simply run
Configure
Make
Make install
Optionally I believe you can run make menuselect first to choose packages?
Hope this
Ciao Andrea,
Hello *,
do queues allow me to set an announce like the A() option of the Dial() cmd?
The announce that I've found is a message that is heard by the caller. I'd
like to send a message to the member of the queue that picks up the call.
In order to help people that find this
Hi
Please don't start three different threads for the same topic,
On Tue, May 29, 2007 at 10:11:44AM -0700, Khaled Chehab wrote:
I am using centos 4.4 server cd
Centos 4.4 or centos5?
,when I am trying to compile zaptel 1.4.2
,the following error appears
From where I can get the
On Tue, May 29, 2007 at 11:53:59AM +0200, Mats Karlsson wrote:
1. Read the error No such file or...
Do a yum install autoconf
Totally unrelated.
And as a sidenote, generally autoconf should never be needed for
building software (unless you patched configure.ac or you checked
it from the
Still not working! Grrr!
[204]
exten = 204,1,Wait()
exten = 204,2,Answer
exten = 204,3,Playback(demo-congrats)
exten = 204,4,Hangup
exten = s,1,Dial,(Zap/g2)
exten = s,2,Hangup
;exten = 204,Voivemail(u100)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi Gilles,
I think that you must be take the incomming call control using AGI perl
scripting. Take a look on this script:
;extensions.conf
[internal]
exten = group,1,LookupCIDName
exten = group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)}|
${EXT204})
=
I am doing something major wrong here. I can not even see it hit the CLI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BSumrall
Sent: Tuesday, May 29, 2007 3:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users]
The priority 2 in extension 204 of context brad is duplicated:
...
exten = 204,2,Answer
...
exten = 204,2,Dial(Zap/g2,20)
...
Sds,
Gustavo
From: BSumrall [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: 'Asterisk
Debug says this???
SIP/2.0 401 Unauthorized
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BSumrall
Sent: Tuesday, May 29, 2007 3:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Trying to dial out on teliax
Here is my basic question which is not answered directly big bird-cookie
monster style in any of the literature.
Sip.conf is your basic authentication file which the CLI: show sip peers
tells you if you are correct. I got that much.
Now, do you have to declare that again anywhere in
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4060e
(gsm|ulaw|alaw|speex|ilbc|h261)/video=0x4 (h261), combined - 0x40e
(gsm|ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP
I was able to catch that.
Here is what I am working with now!
Just in the weeds on this learning curve!
[204]
exten = 204,1,Wait()
exten = 204,2,Answer
exten = 204,3,Playback(demo-congrats)
exten = 204,4,Hangup
exten = s,1,Dial,(teliax)
exten = s,2,Hangup
;exten = 204,Voivemail(u100)
As the error message shows, you have wrongly numbered priorities
[brad]
exten = 204,1,Wait()
exten = 204,2,Answer
exten = 204,3,Playback(demo-congrats)
exten = 204,4,Hangup
exten = 204,2,Dial(Zap/g2,20) --
;exten = 204,Voivemail(u100)
Anyone else with any suggestions?
On 5/28/07, Kapil Dhawan [EMAIL PROTECTED] wrote:
Redhat Enterprise
Zeeshan Zakaria wrote:
I've to setup Asterisk on a Dell PowerEdge 1750 server. Its dual Xeon
3GHz with Hyperthreading. People on this list who have experience with
this server please
Hi
I would like to disable correctly musiconhold for my users when they are
using the callwaiting feature.
I have set in modules.conf
noload = res_musiconhold.so
Now I don't have music on hold when I use call waiting but I have this warning:
-- Music class default requested but no
We are running Asterisk on native CentOS. We then install VMWare on
CentOS with Windows 2003 in the VMWare partition for AD services. We
have 50+ users in a call center environment with no issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
I have address that issue.
I seem to be getting a unknown user error from teliax.
Apparently, I am not sending the call through my authenticated trunk in
sip.conf?
Suggestions anyone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent:
Hello,
I'm thinking of ordering an OpenVox A400P01 (A400P + 1 PORT FXO Bundle)
for use in a old IBM 8364 thin client:
http://www.openvox.com.cn/products_detail.php?genre_id=9id=28
http://silicon-verl.de/home/flo/software/netstation-8364/
Has someone already used this hardware with Asterisk,
On Tue, 29 May 2007, Zeeshan Zakaria wrote:
Anyone else with any suggestions?
Hard to work out what to suggest - what's your expected load going to be?
Any telco cards? etc.
If you want support from Dell, then it's RHEL whatever...
Personally, if I had that hardware, I'd load up Debian
Hi Steve -
It's definitely ftp. I have given the phone a static ip. When I set it to
dhcp it just
hangs and cannot get an IP. I can ping the phone and see the web config page
so it is on the network.
Any more suggestions.
I once saw this same sort of behavior on a Polycom 501. It got
I have this card and no problem.
It is very simple to configure.
Go on!
2007/5/29, Gilles Ganault [EMAIL PROTECTED]:
Hello,
I'm thinking of ordering an OpenVox A400P01 (A400P + 1 PORT FXO
Bundle)
for use in a old IBM 8364 thin client:
No, but I think that you can't install this OpenVox board in this
NetStation case, because the card is a full length PCI and the PC case
supports only half length PCI cards.
Sds,
Gustavo
From: Gilles Ganault [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
On Tue, 2007-05-29 at 08:21 -0400, Zeeshan Zakaria wrote:
Anyone else with any suggestions?
A white back there has been a lot of discussion about Dell servers and
Asterisk. Search the archives.
Regards,
Patrick
On 5/28/07, Kapil Dhawan [EMAIL PROTECTED] wrote:
Redhat Enterprise
Rather hasty I think. I think whatever version 1.2.X winds up on should
be the most stable release of Asterisk, period.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olle
Hi,
Is it possible to implement some kind of alarmist triggering some
action, by sending SIP OPTIONS messages regularly to check that other
peer is still online?
I'm using Asterisk version 1.2.11 which I know it doesn't have any SNMP
module, just 1.4 branch is being developing one; but is it
Steve Totaro wrote:
Rather hasty I think. I think whatever version 1.2.X winds up on should
be the most stable release of Asterisk, period.
Great -- so do I. But I'm not developing Asterisk either.
It was going to happen sooner or later -- at least this will encourage
more people to get into
A common attitude in the development community.
Keep adding more bells and whistles, it's more fun and interesting.
Don't bother to fix the many existing problems. That is boring
Peg Leg O'Brien
Steve Totaro wrote:
Rather hasty I think. I think whatever version 1.2.X winds up on should be the
I have an install using Rhino cards -- I sure hope they get their act
together by then.
on Tuesday 05/29/2007 Stephen Bosch([EMAIL PROTECTED]) wrote
Steve Totaro wrote:
Rather hasty I think. I think whatever version 1.2.X winds up on should
be the most stable release of Asterisk, period.
John Novack wrote:
A common attitude in the development community.
Keep adding more bells and whistles, it's more fun and interesting.
Don't bother to fix the many existing problems. That is boring
You know what's more boring? Having two feature-frozen versions of the
same software fighting
A common misbelief from those who do not code every day or manage large code projects.
When you release a version, you have to fix the feature set. The fact that they have worked on 1.2 so long is a testament to the fact that bug fixes are very satisfying.
Second, 1.4 will have no new features,
FYI,
http://www.voip-info.org/wiki/index.php?page=Asterisk+FAQ
*Can i install Asterisk on a beowulf cluster?* A cluster can't migrate
threads that use shared memory. Asterisk uses that kind of threads.So no,
Asterisk wouldn't work on a cluster. *(It might be helpful to know whether
anyone has a
Hi,
Be careful with believing too much that your zaptel hardware will work
together with xen, you could have problems like the ones described in
the thread linked below:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180825.html
Good luck,
François.
Adam Robins wrote:
We
John covici wrote:
I have an install using Rhino cards -- I sure hope they get their act
together by then.
They have no choice now, do they?
Nothing focuses the attention like a deadline.
-Stephen-
___
--Bandwidth and Colocation provided by
Thanks, but we do not use any zap hardware in these systems. It is straight
SIP.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François
Delawarde
Sent: Tuesday, May 29, 2007 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Khaled Chehab wrote:
I am using centos 4.4 server cd ,when I am trying to compile zaptel
1.4.2 ,the following error appears
From where I can get the missing rpms .or kernel source ,or the kernel
header
Generating input for menuselect ...
grep: /include/linux/autoconf.h: No such file or
Khaled Chehab wrote:
I am using centos 4.4 ,when I am compiling zapltel using l make linux26
,error accrued ,what s missing
[EMAIL PROTECTED] zaptel]# make linux26
grep: /include/linux/autoconf.h: No such file or directory
make: *** No rule to make target `linux26'.
I'm having the same trouble when the Alcatel-Asterisk
trunk has prefix meaning set to open routing
number.
I enabled overlapdial but still get the same behavior.
When it's set to routing number Asterisk receives
the full dialed number but it's limited to a maximum
of 8 digits.
Has anyone solved
I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center
with 6 agents. I am using realtime for queues and sip and I am also
trying to use realtime static to load agents.conf. The only problem I
am having is that no agents are loaded when I start Asterisk. I have to
Thanks for the answer Matthew.
I'm having high load, choppy sound and slow responsives with an
asterisk server (version 1.2.12.1) that make a peak of 90 channels
(around 60 phones calling at max, isn't necessary to reach this peak
to get the problem). All the traffic is SIP, with
On May 25, 2007, at 4:24 PM, William Moore wrote:
On 5/25/07, Alex Balashov [EMAIL PROTECTED] wrote:
In reference to an old post from 2002:
http://www.marko.net/asterisk/archives/0203/0103.html
How does one go about doing this?
I think what mark was referring to there is dynamic spans.
On Tue, 29 May 2007, Matthew Fredrickson wrote:
SS7 support is in trunk right now. We can always use more testers for the
stack :-)
Thank you Matthew. Appreciate it. I don't really have any A-links to
tap right now, but will get on that ASAP.
Does the stack support feature /
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .
On 29/05/07, Stephen Bosch [EMAIL PROTECTED] wrote:
John covici wrote:
I have an install using Rhino cards
When turning of dhcp, dont forget to set all other attributes
manually. Ones that would effect this are
IP Address
Subnet mask
Gateway
boot method tftp/ftp
Server Address
username/password if ftp
vlan
Assuming you are setting a hard IP for the server, if using a url
then donot forget to add
On May 29, 2007, at 11:14 AM, Alex Balashov wrote:
On Tue, 29 May 2007, Matthew Fredrickson wrote:
SS7 support is in trunk right now. We can always use more testers
for the stack :-)
Thank you Matthew. Appreciate it. I don't really have any A-links
to tap right now, but will get on
On Tue, 29 May 2007, Matthew Fredrickson wrote:
Not right now. I'd like to see that done, however, I've been mostly
focusing on making sure that its current feature set is solid.
So it basically just does ISUP?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel
In my experience, many times Qsig is mandatory for interconnection
between Asterisk and others PBX using PRI.
Jorge Mendoza
Vieri wrote:
I'm having the same trouble when the Alcatel-Asterisk
trunk has prefix meaning set to open routing
number.
I enabled overlapdial but still get the same
On Tue, 29 May 2007, Edgar A. Luna Diaz wrote:
The real problem was found. The configuration of this server had a
recording path as /var/spool/asterisk/monitor/ for every call, so the
size of monitor (the directory) keeps growing at 2000 files per day. Its
peek was around 36MB, just containing
Hi,
would it be possible to use Asterisk to record calls only? There would
be an existing PBX and calls come in on a ISDN30 line?
The Asterisk box would need to sit between the incoming ISDN 30
circuit and the existing PBX.
Is this possible?
thanks
Mike
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .
Fair indeed. I would guess that a completely stable 1.2 w/ security
maintenance is acceptable to the
According to
http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI
the author had trouble with QSIG.
It would be great if you could give me an extract of
your zapata.conf in your successful QSIG setup. And
any other tip for that matter.
--- Jorge Mendoza [EMAIL PROTECTED] wrote:
In my
Mike,
This thread might be of aid:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180994.html
-- Alex
On Tue, 29 May 2007, Mike Dent wrote:
Hi,
would it be possible to use Asterisk to record calls only? There would
be an existing PBX and calls come in on a ISDN30 line?
The
Sahil,
Start here,
http://www.alcatelunleashed.com/viewtopic.php?f=44t=840
http://www.alcatelunleashed.com/viewtopic.php?f=44p=8595
John Treble
Ottawa, Canada
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Vieri
Sent: May 29,
Keep in mynd, SIP requires a stable timing source. Don't know how Xen handles
timing, but with vmware you can get all sorts of issues with timing: the clock
goes faster or slower then normal on multi core systems and on systems with
power stepping.
In my case i'm getting those timing issues
Michael Collins wrote:
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .
Fair indeed. I would guess that a completely stable 1.2 w/ security
maintenance is
Hi,
This contacted call has no audio, any ideas?
The conference suite from another provider on internal IP is waiting
for an ACK on port 5605, but * is sending it back to port 2289
Internal between Asterisk and another Conference suite:
* SIP Call
Direction: Outgoing
Call-ID:
This is why we installed Asterisk on CentOS directly and then put Windows under
a VMWare partition, rather than put bot CentOS and Windows under VMWare
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund
Sent: Tuesday, May 29, 2007 1:59 PM
To:
I haven't used the iaxmodem / hylafax combo for sending, only for receiving.
However my experience is that it is 99% reliable. I am using a Dell
PowerEdge 850 with a Pentium 2.8Ghz and 512mb ram. I think it is the
Pentium D but could be the dual core, not sure, whatever the base cpu was at
the
Thanks Jerry,
The dns seemed to be the culprit
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Tuesday, May 29, 2007 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Static
Mike,
First, what kind of T1 card(s) do you have? (Just curious.)
I've seen two different theories of operation, although I have
experience only with one, and that's not with Asterisk. One is a
passive tap, the other is a pass-through. I can't say that I know if
they work or how, but
Except that for some users 1.2.18 is NOT stable. I've had to roll
back
to 1.2.15 on my production servers in order to prevent core dumps at
least once per day. No, I am not willing to turn my production
servers
into testing servers to solve this. Doing so would make me a former
consultant
On 5/28/07, shadowym [EMAIL PROTECTED] wrote:
Thanks for all the replies. Seems there are at least 2 or 3 people giving
strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade
production) solution. That is just the sort of feedback I was looking for.
My application is
In that case your post about vmware has nothing to do with the problem
discussed here.
Adam Robins wrote:
Thanks, but we do not use any zap hardware in these systems. It is straight
SIP.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François
Hello list,
I have a question here that may be a little bit strange for some of you.
I would like to send an INVITE from Asterisk to a given client
without any SDP anouncement in it. Indeed, that is pretty useful for
Click to call applications for instance, where you have no way to
know
Well i guess you just need a good look on logs for why and when you
are getting core dumps . We are having few servers running .1.2.18 and
it has turned out to be most stable in whole 1.2 branch ( had some
issues with 1.2.13 and 14 ) .
Except that for some users 1.2.18 is NOT stable. I've had
Well, yeah, I know it's do-able with either the Sangoma card or Digium's own
TDM2400 but I don't want to replace the TDM400p I've already got in
there
Anyone think two TDM400p's won't cause me any trouble?
--
Chris
Lee Jenkins [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Eric ManxPower Wieling wrote:
Michael Collins wro
Except that for some users 1.2.18 is NOT stable. I've had to roll
back to 1.2.15 on my production servers in order to prevent core dumps
at least once per day. No, I am not willing to turn my production
servers into testing servers to solve
Agreed.. I think we need to freeze 1.2.x in a stable release.. right now
there are too many bugs in it to be considered stable.
On 5/29/07, Steve Totaro [EMAIL PROTECTED] wrote:
Rather hasty I think. I think whatever version 1.2.X winds up on should
be the most stable release of Asterisk,
Which sounds like exactly what I described. Asterisk in Dom0...
-Jonathan
Adam Robins wrote:
We are running Asterisk on native CentOS. We then install VMWare on
CentOS with Windows 2003 in the VMWare partition for AD services. We
have 50+ users in a call center environment with no issues.
Well, asterisk was working without flaws until just a few minutes ago.
Asterisk stayed running, but we were just getting dead air when you'd
pick up the phone and you defiantly couldn't send/receive calls. I
looked at the call detail, and there isn't anything abnormal there. I
then looked at the
Hi,
Except that for some users 1.2.18 is NOT stable. I've had to roll back
to 1.2.15 on my production servers in order to prevent core dumps at
least once per day. No, I am not willing to turn my production servers
into testing servers to solve this. Doing so would make me a former
consultant
I've got an open bug regarding version 1.4. When I tried to find a
'stable' version I tried 1.2.18, it had the same problem. I then tried
1.2.13 and the server hasn't had an issue since.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
What you say might be true for small business or home pbx systems .
But if you have a production server handling sip/iax trunks over
internet then you need to upgrade to avoid security related bugs and
exploits that are released .
You seem to miss the idea here. You work with a version
I am interested in doing some hotel installs of asterisk, but I too am
confronted wtih the hurdle of making it play nice with reservation systems
such as Micros System's.
Have you made and discoveries?
- Mark
___
--Bandwidth and Colocation provided by
On May 29, 2007, at 11:54 AM, Alex Balashov wrote:
On Tue, 29 May 2007, Matthew Fredrickson wrote:
Not right now. I'd like to see that done, however, I've been mostly
focusing on making sure that its current feature set is solid.
So it basically just does ISUP?
Yes. :-)
Matthew
Have you made any progress. I am interested in the same thing.
- Mark
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I am trying to make a mirroring for my asterisk using nextone SBC,I have a
problem ,which is when and end point send Invitation to SBC realm .
This realm is send INV and REG messages to Asterisk. Asterisk sends INV
message again to this realm.
NexTone SBC try to send again to asterisk
On 5/29/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center
with 6 agents. I am using realtime for queues and sip and I am also
trying to use realtime static to load agents.conf. The only problem I
am having is that no
Eric ManxPower Wieling escreveu:
Michael Collins wrote:
Except that for some users 1.2.18 is NOT stable. I've had to roll back
to 1.2.15 on my production servers in order to prevent core dumps at
least once per day. No, I am not willing to turn my production servers
into testing servers to
On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote:
Michael Collins wrote:
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .
Fair
Hi
i have 20 people calling agents calling
when ever they calling i get this below error
May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided
deadlock for '0x8b2f50', 10 retries!
and the voice go choppy, and voice breakages
iam using Latest SVN, any suggestion to come
Well if you are out of luck with asterisk .. How about its fork
callweaver ? I am highly awaiting its stable release to see if it
holds upto what its wiki says .
On 30/05/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote:
Michael
Is it over iax and there are lot of outgoing channels ? If yes then
you are not the only person having this ..
On 30/05/07, ram [EMAIL PROTECTED] wrote:
Hi
i have 20 people calling agents calling
when ever they calling i get this below error
May 30 00:46:57 WARNING[2840]: channel.c:785
At 16:17 5/29/2007, Tzafrir Cohen wrote:
On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote:
Michael Collins wrote:
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate
I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card
and an OpenVox A1200P card. Up to today everything was working
perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo
ports are used for a GSM adapter and for an ATA connected to Vonage.
The
On Wed, May 30, 2007 at 03:35:36AM +0530, Jaswinder Singh wrote:
Well if you are out of luck with asterisk .. How about its fork
callweaver
Is it not enough that they have forked Asterisk, you now want to fork
from CallWeaver? You're sure after some interesting life ;-)
? I am highly
On 5/30/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
Is it over iax and there are lot of outgoing channels ? If yes then
you are not the only person having this ..
SIP
ram
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On Tue, 2007-05-29 at 14:51 +1200, Carlos Hernandez wrote:
Please get in touch off list.. We're wanting to hire a professional
subcontractor, developer or company to get around some issues like these:
Please let me know if you have done this type of work before. We are not
wanting to
On Tue, May 29, 2007 at 05:22:45PM -0500, Carlos Chavez wrote:
I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card
and an OpenVox A1200P card. Up to today everything was working
perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo
ports are used for a
On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote:
What is the output of:
show channels
Any chance that there is actually another call that keeps that channel
busy?
No, the line is not busy with another call.
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