Hi,
SpanDSP works great for me, even over SIP.
I did change the fax macro a bit to make it work better.
The macro records total number of faxes and pages received by the
extension.
Here is the example of 7644 fax extension.
1. Have this in your dialplan
exten = _7644,1,Macro(faxreceive)
I made the same mistake with my 7960
The content of 'OS79XX.TXT' should be P0S3-07-4-00 and not P003-07-4-00
Same goes for SIPdefault.cnf.
After the change everything worked like magic
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Delete /usr/lib/asterisk/modules/app_md5.so and update your dialplan if you use
MD5.
It is now done in functions.
/usr/lib/asterisk/modules/app_md5.so is a leftover from your previous
installation.
[app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource:
I guess I just assumed that that the connection to asterisk would have
to be IP since it is absolutely impossible to connect ~208 T1s directly
to a single asterisk server. You would have to use an external media
gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :)
Not
The main sever is still connected via IP, correct?
Does not matter if you use * for media gateways or an APX8000 - the
only
trunking options to get to the main box are IP based.
Are seriously going to tell me that a quad xeon/opteron would not handle
traffic from 4xGIG cards?? :)
Hi,
Has anyone noticed degraded voice quality with HPEC?
I have a client running TE4XX card who configured HPEC for couple of
channels with echocancel=1024.
Whenever HPEC is used you get a background static in voice.
When HPEC is not used everything is crystal clear.
What could cause this
Performance Echo Canceller (HPEC)
In article
[EMAIL PROTECTED],
Boris Bakchiev [EMAIL PROTECTED] wrote:
Hi,
Has anyone noticed degraded voice quality with HPEC?
I have a client running TE4XX card who configured HPEC for couple of
channels with echocancel=1024.
Whenever HPEC is used you get
Hi,
I'm unable to load wct4xxp module for TE406P card.
I've compiled 2.6.13-rc6, got latest CVS (as of today) of zaptel, but
when I try to load the module I get this:
kobject_register failed for Unified t4xxp/t2xxp driver (-13)
[kobject_register+53/73] kobject_register+0x35/0x49
It works out that name Unified t4xxp/t2xxp driver is not accepted
anymore by 2.6.13 kernel.
Need to remove / for it to load properly
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: Monday, 15 August 2005 18:17
2n also asked to buy from local
distributor, and Im glad they asked.
Works out that here in Australia distributor had to get
the units tested and certified for us to be able to use them.
I just checked the 2ns site, all
downloads are available without logon for VoiceBlue products..
Hi Raph,
We have bought the units from the same supplier as you (Talk to Us).
All our calls take about 5 seconds before the mobile we're calling
starts ringing. Some calls take up to 7 seconds but I think it depends
on the carrier.
I believe we have even tested one of your units for Matt because
Hi,
Did anyone compare G.729 implementations (from Digium and the one based
on IPP) on features, stability, quality and reliabilty?
It would be intresting to know how they fair against each other.
I could be wrong, but in my testing I did notice a bit more hiss on
Digiums codec
Rod,
Here is my macro for this:
[macro-sipexten]
exten = a,1,VoicemailMain(${ARG1})
exten = a,2,Hangup()
exten = s,1,DBGet(NATIMEOUT=Features/${EXTEN}/NATIMEOUT)
exten = s,2,Dial(${ARG2},${NATIMEOUT})
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,102,Goto(s,350)
exten =
Hi,
Wondering if there is a timer provided on TDM cards?
I don't have use for TE110P and it seems expensive just to get it for
timer function.
I do have ztdummy running but it is hovering on 99.975586% and I'm not
sure if this is good enough or not.
Any info is appreciated.
This message
We're running asterisk on a pair of 1GB 12mb/s flash cards running on
separate IDE channels.
We've setup software RAID1 to protect ourselves from failures if any of
the flash cards die.
VoiceMail is stored on a small IDE that is dedicated just for this.
It appears to work quite well. Although
Most probably your server was busy starting up when asterisk loaded and
calculated the table.
Next time, just issue show translation recalc without after the server
settles down.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Irakli
I would use g.729, and if this is an issue, GSM.
Setup trunking between both IAX peers so that you can save a lot of
bandwidth.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, 4 May 2005 00:52
To:
I had CC readers going over the internet (with pings over 80ms)
connected to Linksys PAP2.
It was only successful once every 3 attempts.
I had 100% reliability when it was connected on LAN.
Timing is an issue, if you doing everything on LAN it should not be a
problem. Just make sure you use G.711
Steve,
Do you know if Digium's implementation has any of those features?
I was not able to find any tech info about it.
There are Annexes up to I. The earlier versions are fixed point,
reduced
complexity fixed point and floating point at 8kbps. These are all
compatible. Later annexes add
Hi
Can TDM400P detect polarity reversal on FXO module?
We have C.O. lines that reverse polarity on Answer and release.
Thank You
This message (and any associated files) is intended only for the use of the
individual or entity to which it is addressed and may contain information that
is
Yes you can.
Just tell iax to bind to that virtual address in iax.conf
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lance Grover
Sent: Thursday, 16 June 2005 14:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
In article
[EMAIL PROTECTED],
Boris Bakchiev [EMAIL PROTECTED] wrote:
Yes you can.
Just tell iax to bind to that virtual address in iax.conf
I don't think that will work on the box that doesn't currently own
that virtual address.
I think the only way is to make sure the bind address is 0.0.0.0
The registry's are stored in DB.
Just export your database with 'database show'
Schedule it with cron to run every 5 minutes or so.
You can do that with -rx command line switch for asterisk.
Send the file across to other node and pipe it through awk/perl/cut or
whatever you like and import it
This doesn't answer the original question - why do you need to reload
it?
I'll give you an example.
An Active-Active asterisk cluster.
In the event one of the servers dies, the other server can take over
without loosing registrations. Since most of the SIP clients know how to
use DNS
Most of the problems like these for me are gone since I started using
iaxmodem+hylafax combination.
Hylafax has ECM capability which just tells the other side to resend the
affected frames (not the whole page).
With the latest 4.2.5.5 hylafax I even have color support :) Not that I
probably
That's not entirely correct :)
Fax and voice on the same DID is not possible when using a second
application like hylafax. Because how should the two applications
decide
which one accepts the call?
With the help of iaxmodem (which works really well) its easily done!
Just detect the incoming
I Guess you can edit the following line in your hylafax config file for
your iaxmodems.
Class1RMQueryCmd: !24,48,72,96
Put exclamation in front of 96 (as it is done with 24) and it should
disable the receive with that speed.
Is there a way to limit the speed of Hylafax to 7200
The simplest solution and the one already implemented in linux, tmpfs.
It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel
do the work it was designed to do. And you would not be limited to PCI
bus speeds. The DDR2800 is about 12GB/sec. Some would say overheads,
etc, etc.
Is this dual CPU/Core or just P4 with HT enabled?
If it is P4, I would recommend to disable HT.
Try changing PCI slots for one of the cards (if you have spare PCI slots).
CPU0 CPU1
0: 17697848 17714488IO-APIC-edge timer
___
Are there any advantages/disadvantages to using tmpfs as opposed to
the
following method:
Matt,
Its simple. To quote the docs, tmpfs lives entirely in the kernel's
caches
It will shrink and grow to accommodate the files that currently on the
filesystem.
So if you allocate 10GB for your /tmp
Our production asterisk server has TE411P and we route close to 50-70K
of calls per month through its ports.
We have NEVER EVER had any issues with faxing (close to 3k/month) with
faxes connected on one of the spans of the card.
Moreover, we have had quite a success receiving the faxes with
Opened pseudo zap interface, measuring accuracy...
99.987793% 100.00% 100.00% 100.00% 100.00% 100.00%
100.00%
100.00% 100.00% 99.987793% 100.00% 100.00% 100.00%
100.00% 100.00%
100.00% 100.00% 100.00% 100.00%
I had the same problem!
You have in your PXXX in your configs that 1.1.0.11 does not support.
Took me an hour to go through my configs and the web page to find what
PXXX in my configs unset the phone :)
Once its done, the phone will be accept the configs with no problems.
-Original
HI Ben,
Make following context in your extensions.conf
[notifycallrec]
exten = tone,1,Answer
exten = tone,2,Answer
exten = tone,3,Playtones(!950/50,0)
exten = tone,4,Wait(10)
exten = tone,5,Goto(3)
exten = h,1,StopPlaytones
Then you can call it with:
exten =
HI,
Does anyone know if there is a PCI-X 4 port PRI cards available on the
market?
If so, have anyone used it and how reliable they were?
Any help is appreciated...
___
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Asterisk-Users mailing
Samsung PABX?
Its TEPRI probably configured in overlap mode so you need to configure
asterisk span that is connected to PABX to overlap mode as well.
When user selects the outside line in overlap mode PABX connects to
asterisk and then sends the digits to it as the user presses the key's.
If
These days you don't have to worry much about your write cache unless
you're running application where once single byte changed will affect
whole file.
Look at it this way, the only corruption will occur is whatever the
files were open by asterisk at the time of the crash. And only up to the
The cold hard truth is that if Asterisk cannot achieve 99.999% uptime
without becoming much more expensive that a traditional PBX then it is
not
a
viable alternative. Even elcheapo Key systems are rated for five
nines.
That is what the telco world requires unless your just using Asterisk
in
Its slow :) It will give you some delays but it will not be noticeable
(most voice files are 5-100kb, so it should be ok... But writing to
them.. Not sure.. It should be ok as well I'm guessing as kernel will
provide some caching (since you have G and not GS it has less ram, so
maybe chaching is
Hi,
We do J
We use iaxmodem+hylafax combo on TE406P
card.
Around 4K of faxes were received without
any problems (some faxes are over 80 pages long!)
It is working really well!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pibix
Sent: Friday, 16
Hi,
I have TE406P (2nd gen card with echo cancellation on-board).
We still notice quite often echo on our PBX that is connected to one of
the spans on TE406P (with calls routers to PRI provider on another
span).
I've tried to experiment with the echo cancellation on asterisk.
I enabled echo
Hi Kevin,
Thanks for your reply.
That probably what it was. :)
Could echo cancellation on PBX conflict with VPM module and create the
warping babble sound that my users are reporting?
Do echocancelwhenbridged and echotraining do anything when VPM module is
used? Should I be using them?
Regards
Hi,
Is anyone running zaptels watchdog in production?
Any adverse effects on using it?
Thanks
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Asterisk-Users@lists.digium.com
Get VoiceBlue VoIP GSM gateway.
It works very well with asterisk.
I have been using it for the last 4 month and its fantastic!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Friday, October 28, 2005 10:27 PM
To: Asterisk Users
Hi,
Im getting a lot of false DTMF detections on my
system.
Following is a diagram of my system:
PRI-TE406P SPAN1-TE406P SPAN3-PABX
Basically anyone talking to me with a higher pitch
voice (Ladies) I get beeps all over the place.
If I unplug PRI from Asterisk and plug it
You can do multiple g723 codecs on PAP2 though.
Yeah, I can confirm that. I added more allow
statements for other codecs for that device as a
fallback. Either codec works great, just not at the
same time when calling each other.
___
--Bandwidth
I think someone needs to start some sort
of wiki that everyone can enter the details of they systems J
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of vador loupe
Sent: Thursday, 17 November 2005
09:54
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Hi,
I have a peculiar problem with asterisk using 100% cpu (one of the
thread just nails one of the CPU's on dual-code system).
Asterisk is running chrooted and under its own username.
If I alter the init script and add -c to PARAMS variable one of the
CPU's is being hammered by asterisk.
I
Hi
Might check to see how many mpg processes are running, or use top to
see if that's the culprit. If so, kill off the mpg that's doing it.
I'm not running any mpg123 processes as I'm using native music on hold
(raw files)
It has something to do with the color option for the asterisk.
If I
Hi,
What would be a recommended PCI latency timing for server running TE406P
card?
Thanks
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Hi,
How would one activate/deactivate hardware echo cancellation on the
TE406 card?
Can it be done per channel?
I'm going to run TE406 in the following scenario:
ISDN - TE406 - PABX
I understand from Steve Underwood's site that echo cancellation is not
good for faxes (and they do that
?
Sep 8 10:36:07 WARNING[13375]: chan_zap.c:8701
pri_dchannel: Hangup on bad channel 0/14 on span 1
-- B-channel 0/14 restarted on span 1
Boris Bakchiev
Jildent Pty Ltd
Tel: + 61 3 8080 5898
Fax: +61 3 9811 4716
___
--Bandwidth
Hi,
I would like to utilise immediate=yes to monimise the delay that simle
switch introduces.
When I set this option, ${EXTEN} is not populated so I Im
unable to do some prepocessing of calls.
Is there a way to populate or retreive EXTEN from a channel thats
been setup with
What kernel are you using?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 7:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TE406p no interrupts
Hi,
I've installed an TE406p, asterisk1.2 on
Well.
Try this please (but only if you're running on the latest sources).
Open wct4xxp.c sources and search for pci_module_init
Replace it with pci_register_driver
So the line should read:
res = pci_register_driver(t4_driver);
That allows you to get the card working on 2.6.13 in almost exactly
and motherboard?
Thank you Boris.
--- Boris Bakchiev [EMAIL PROTECTED] wrote:
Well.
Try this please (but only if you're running on the
latest sources).
Open wct4xxp.c sources and search for
pci_module_init
Replace it with pci_register_driver
So the line should read:
res
Well.
That means pci_register_driver probably not ding what it supposed to do.
In newer kernels pci_module_init should be replaced with
pci_register_driver as pci_module_init doesn't it what it supposed to.
How brave are you at getting a new kernel on your system?
I'm currently running on 2.6.13
I'll vouch for them. :)
Very nice people and service.
Boris
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Callum McGillivray
Sent: Wednesday, 14 September 2005 12:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
My output from TE406P is:
483328 samples in 60.415876 sec. (483327 sample intervals) 99.999794%
483328 samples in 60.415900 sec. (483328 sample intervals) 100.00%
483328 samples in 60.415872 sec. (483327 sample intervals) 99.999794%
Estimate 8 frame slips every 483.328003 seconds.
Does
HI,
How many frame slips would spandsp tolerate before faxing becomes
impossible?
Using ztclock my current system slips a frame every 60
seconds.
Does each frame slip means a failed fax or will there be retransmission
of the block/page that had the frame slip?
Regards
Boris
Hi Kris,
I have TE406P (same as your but quad span) working on 2.6.13 with
pre-empt.
I had it working fine with 2.6.14 but I could not switch card's IRQ from
CPU0 to CPU1 on the 2.6.14
On 2.6.13 CPU1 is handling IRQ's only for TE406P (with occasional timer
IRQ's sneaking in).
I suggest that you
I like the chan_ooh323.
I like the idea of selfcontained H323 channel that doesn't rely external
libraries, often with specific versions that conflict with something
else.
OOH323 works right out of box and since we started using it to
interconnect Asterisk to Samsung OfficeServ 500 we had no
Dec 2005, Boris Bakchiev wrote:
I like the chan_ooh323.
I like the idea of selfcontained H323 channel that doesn't rely
external
libraries, often with specific versions that conflict with something
___
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In software asterisk can support more than that, no?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Tuesday, 6 December 2005 17:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo
Why not?
Digium works hard in hardware software department.
It constantly improves its hardware offering.
The software arm has been busier then ever! Million bug fixes, MANY MANY
improvements, roadmap (at least from what I can see from contributing
developers in SVN) is amazing.
Asterisk and
Can the TE406P card's VPM module be swapped for the new revision with
Octasic chipset?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Sunday, June 25, 2006 8:08 PM
To: Asterisk Users Mailing List -
Anyone knows how to contact maintainers of Chan_gsm_bt?
They http://changsmbt.free.fr/ site has no contact details.
I believe I found the issue why it does not initiate SCO links
properly..
It looks to be a timing issue. It sends additional AT commands without
waiting for the responses for
Hi,
I have a SIP provider who sometimes sends duplicate RTP packets to me.
Sent RTP packet to 10.55.20.201:17440 (type 08, seq 008536, ts
4846560, len 000160)
Got RTP packet from10.55.20.201:17440 (type 08, seq 051978, ts
3647104992, len 000160)
Got RTP packet from
Replying to myself
Its fixed now
Checking timestamps is optional according to RFC so asterisk is not
doing it.
Anyway, I made a patch and tested it and its working.
Thanks.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris
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