hi
Not all E1 providers have crc4 turned on.
Except for ISDN lines, E1s hardly ever have CRC4 turned on. As you
said, some countries don't even turn it on for ISDN, which is stupid.
It is a historic reason for this cause the first E1/ISDN connections
where on computers that did not
Hello,
I am considering using Asterisk to replace my organizations dying
phone system, and so far everything seems perfect. One question
though, our Executive Director has gotten into the habit of using our
existing system to leave voicemails for multiple recipients. For
example, he
The RTP packet size defaults to .03 (packet size in seconds). Changing this
to .02 or .01 fixed the issue with Meetme. Anything .03 or above introduces
the doppler effect in a Meetme conference. Thanks. Codec is uLaw and
silence suppression was off already.
Now, however, there is a (very)
I have one SIP phone (and soon a 2nd phone) and a Digium TDM11B (1 FXO +
1 FXS) card. I would like to be able to dial out the analog line via
Asterisk. How do I configure that?
i.e I'd like any extension to be able to dial 411, 911, 0, (617)
555-1212, 16175551212, etc... and have these
Robert Webb wrote:
Hi all, hope this is not too off topic, but thought it fit better here
than any of the other lists.
I am in the IT field with some telecommunications background
knowledge. Mainly learning from when my father worked for a telco and
from keeping up the old Nortel PBX here
Cool application, one question, can you call from any phone? Is the
number on the reference card an example, customer support number, or the
number you call into to login? Does the system only call you or is there
a way to call in and check the status?
Thanks,
James
Chris Tooley wrote:
The number on the reference card is the caller id that your auction call
will come from. It's an outbound only application at the moment. We'll
be adding inbound at some point in the future, but we need to stabilize
this product first. There is a simulation application that you can have
call
Yes, you can forward voicemails or broadcast them as well.
OT: What does the Marijuana Policy Project do?
Thanks,
Steve
Hello,
I am considering using Asterisk to replace my organizations dying
phone system, and so far everything seems perfect. One question
though, our Executive
Do you have canreinvite=yes? If you do change it to no. If that works
then read the rest of this thread for options if you do not want all
streams to through asterisk.
Thanks,
Steve
I have a related issue.
I have everything set up correctly so that I CAN use live recording
(Press *1 to
I think Ryan is right.
We have been using Realtime + mysql. We also run reports in realtime
that query the mysql db, particularly the cdr table. Al this runs in
realtime, which is very important for reporting. It not very cpu, mem
intensive.
yusuf
This sounds like a prime candidate for
Very neat, useful and extremely creative idea. I wish you much luck
with this product. One question I have and didn't see it documented.
What does the Right Price feature do? Also, can bids be scheduled for
a certain time, like 3 seconds before the close of the auction?
I could see Ebay buying
Here is an example, very basic.
sip.conf
=
[1000];assuming that your first SIP extension is 1000
... ;all the other paramters
context=internal ;default context is [internal]
extensions.conf
[globals]
OUTBOUNDCHANNEL=Zap/2
[internal]
include =
Hello everyone,
AstLinux 0.3.0 has been released. This is the first stable release of
AstLinux to include Asterisk 1.2. There are (of course) numerous other
fixes, more rc.conf variables, a kernel upgrade and some other goodies
as well.
AstLinux is now available in the following
wat does 'show hints' say?
asterisk*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
101 : SIP/101SIP/102 State:Idle
Watchers 0
100 : SIP/100SIP/102 State:Idle
Watchers 1
Ivan Lopez wrote:
/ -I have
When I try to compile app_rxfax.c app_txfax.c I get a big list of the same
error messages. (like: but no such parameter)
I solved the same problem with a previous installation but I don't how...
stupid!!!
Makefile:
all: app_rxfax.so app_txfax.so
app_rxfax.so: app_rxfax.c
gcc -shared
after cvs update i recompiled asterisk now i get this on loading:
Dec 8 20:15:54 VERBOSE[25425] logger.c: [app_md5.so]Dec 8 20:15:54
WARNING[25425] loader.c: /usr/lib/asterisk/modules/app_md5.so: undefined
symbol: option_priority_jumping
Dec 8 20:15:54 WARNING[25425] loader.c: Loading
I'm not sure if Priority is the correct term, but it is the order number
as in exten = fax,1-
If I have an application that loads / includes another file, will a line
of the same order in the included file override the one in the main
application? What I need to do is:
[test]
include
There's automatic snipe bidding, eBay wouldn't let us be an affiliate if
we did that. However, you can enter a max bid and hold off until the
last few seconds to hit the confirm button.
RightPrice lets you enter a price for an item and you don't get the call
if the item's price exceeds the
Has anyone confirmed this? It sounds like an interesting theory.
- Waldo
On Dec 8, 2005, at 12:46 PM, Philipp von Klitzing wrote:
Hi!
This and Time Bandit's comment makes sense. I didn't realize that
these options in the Dial string will force Asterisk to stay in the
media path even if
Has anyone created a call screening script (AGI or other)? What I'm looking
for is for Asterisk to:
1. Pickup line
2. Play Record your name to be connected
3. Call an extension
4. Play the recorded name
5. Prompt 1 to accept to 2 to refuse
6. Either connect the call if 1 or send to VM if 2.
Steve,
Thanks for the quick response. Just to clarify, can this be done
over the phone, or does this require a web interface? Our ED is
often on the road, so he could be doing this via his cell phone.
What does the broadcasting feature do?
MPP is an advocacy group focusing on reducing
On Thu, 2005-12-08 at 08:47 +, David Waugh wrote:
Hello Patrick,
I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with
chan_capi_cm.
I am able to do ISDN to SIP calls with this.
Have you tried using the Eicon drivers instead, rather than zaptel and zib
pri.
Ryan Booz [EMAIL PROTECTED] writes:
Now, however, there is a (very) slight echo introduced into any calls made
to this extension. So obviously the way that the phone sends packets is
causing some issues. Anyone have a resource or guide to point me to on best
way to debug packet transmission
Hi,
I upgraded my chan-ooh323
Same problem
I was running 0.2, now 0.3 (that was the
latest I did found)
Do I need to upgrade asterisk too ?
Up to 1.2.1 ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: jeudi 8 décembre 2005 18:14
To:
I thought the version was up to.7 or .8. In any case I am using the
version in Asterisk-Addons
1.2.0.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jacobso1
Sent: Thursday, December 08, 2005 1:37 PM
To: 'Asterisk
I am having the same issue. There was a patch put in that is supposed
to rewite a blank context to default, but it looks like in the process
this patch has killed the realtime variable passed to the query.
-Jon
C F wrote:
Voicemail in itself does not hangup, * will bring you back to the
Noah Silverman wrote:
Moj,
It is set as the default. *1
When I dial *1 I actually see user pressed *1 to start recording.
I then hear a beep.
The system DOES create in and out files and then combines them to a
single file when the call is done.
The problem is that the file is
It might be. I'm going to work with one of the remote users again tomorrow
to see if we can get it working better. You're also right that the PSTN
calls don't hear the echo, INSTEAD I hear a faint static/waves on a beach
sound whenever I talk though a PSTN set through the system to this user.
I have an AGI script but goal was little different.
But my script can do this by tweaking little bit.
I have a two tables,
caller_numbers and statuses
every new number is logged in caller_numbers.
If that caller call next time, based on the status call can be route to
1. Voicemail
2. Hangup
3.
What codec are you using?
On 12/8/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Noah Silverman wrote:
Moj,
It is set as the default. *1
When I dial *1 I actually see user pressed *1 to start recording.
I then hear a beep.
The system DOES create in and out files and then combines
Yeah, it's been on this list at least once a month, it's on the wiki
as well, as well as on the bug tracker. Google is your friend here.
The follwing link has one example:
http://bugs.digium.com/view.php?id=5574
On 12/8/05, Technical Support [EMAIL PROTECTED] wrote:
Has anyone created a call
I have no idea. Whatever was the default when I set up the system
months ago...
-N
On Dec 8, 2005, at 2:36 PM, C F wrote:
What codec are you using?
On 12/8/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Noah Silverman wrote:
Moj,
It is set as the default. *1
When I dial *1 I actually
Joe, are you running PRI to your Opt 11? I have a
61 and I can;t get my d-channel to come up to save my
life!
From: Joe Pukepail [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 08, 2005 9:21 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users]
Hi list,
Im trying to build a system here at the office and i was thinking of buying this equipment
-- 1 TDM22B
- 1HandyTone 286
- 1Linksys Sipura 841
- 1Linksys Sipura 2100
But I need to buy it from a reseller that ship to Mexico (and includes exportation taxes) or sells in Mexico.
The thing is
[EMAIL PROTECTED] wrote:
This sounds like a prime candidate for a database implementation. That way
you can get very near real-time stats without the overhead of frequent
cronjobs or polling. You number crunching computer would then just grab
the data and crunch away. I'm just now getting
Thank you, Carlos. With the suggestion of
doing the vv, I was able to see exactly what files (upper lower case) it was
looking for, corrected that and now the phone does the updates as it is
supposed to.
Thanks very much.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
C F wrote:
What codec are you using?
ulaw
I don't see what the codec would have to do with this though.
On 12/8/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Noah Silverman wrote:
Moj,
It is set as the default. *1
When I dial *1 I actually see user pressed *1 to start recording.
I
Can someone tell me when SRV lookups are going to be fully supported in
Asterisk? I see we just had a new release, 1.2.1. Considering this lack of
functionality is a huge gaping hole for reliability, I would have thought 1.2.1
would have been a good time to implement this. It's way overdue.
Dan,
The version from objective systems is indeed 0.8
This is a general h323 stack (or driver)
The latest asterisk-specific version (named asterisk-ooh323c) I found is 0.3
From http://ftp.digium.com/pub/asterisk/h323/ there is 0.2
Since I did 'stabilize' my cvs, I just wanted to add a h323
Based on the date, I'd guess that version .2/.3 of the channel
driver has the fixes I needed, so maybe it is a different issue.
I'd recommend reporting the problem along with a debug log and
traces from the router to: ooh323c-devel@lists.sourceforge.net
The developers have been quite responsive
Well, it might not have a translation path available, thats why I
asked to make sure that you are using alaw or ulaw.
On 12/8/05, Darrick Hartman [EMAIL PROTECTED] wrote:
C F wrote:
What codec are you using?
ulaw
I don't see what the codec would have to do with this though.
On
I have some really weird happening with my asterisk setup. I have a
TNT-MAX taking call and forwarding via SIP to an asterisk box (box A).
That asterisk box (box A) forwards the call to another asterisk box (box
B) via iax2. Box B handles all the calls and voice mail and stuff.
When I call
It might be. I'm going to work with one of the remote users again
tomorrow
to see if we can get it working better. You're also right that the
PSTN
calls don't hear the echo, INSTEAD I hear a faint static/waves on a
beach
sound whenever I talk though a PSTN set through the system to this
Hi:
I try to understand why the following message appears.
Does anybody know what Channel 'IAX2/' unable to transfer mean?
Thanks
Miguel
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To UNSUBSCRIBE or
They neglected to return my email long (in Internet time) before I mentioned it here. If they did cancel my account, its not hurting my feelings any. I'm satisfied with my new carrier. More likely that they aren't going to waste their time on a customer they don't perceive as being a commercial
I'm having a strange problem with an analog line connected to an
Adtran Channel Bank. It seems as tho, I cannot make outgoing calls
out of the PBX the analog lines are connected to. I'll explain...
The channel bank has a few analog lines (loop start) coming in to the
FXO cards from a
i got this same error/problem on the latest model of the pundit's with the same digium cardon inspection of the logs (using dmesg on ubuntu) all i saw was 'TDM Master PCI abort' spewed out repeatedly. on inspection using top, i saw it was mostly the log daemons eating cpu cycles tryin to keep pace
We are trying to use Asterisk to set up a call between two SIP devices
and then step out of the path.
- all systems have public IP addresses (no firewalls, no NAT).
- sip.conf has canreinvite=yes for both devices
- ulaw is the only permitted codec so we do not have transcoding issues
(and a
Hitting 0 when asterisk announces that the call has gone to voicemail immediately goes to the 'o' reserved extension:
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMail
On voicemail main/comedian mail (reading your own voicemail box), press
3 for advanced options, then press 4 to place an
What does your dial command look like?
If you have Tt, wW, or hH, then asterisk will always stay in the path.
On 12/8/05, George Pajari [EMAIL PROTECTED] wrote:
We are trying to use Asterisk to set up a call between two SIP devices
and then step out of the path.
- all systems have public IP
oh323 and ooh323, which one would be better for asterisk
On 12/9/05, Dan Austin [EMAIL PROTECTED] wrote:
Based on the date, I'd guess that version .2/.3 of the channeldriver has the fixes I needed, so maybe it is a different issue.
I'd recommend reporting the problem along with a debug log
I'm running asterisk on FC4. All works fine, including musiconhold.
I tried installing ztdummy as directed, since the documentation
indicates that ztdummy is required for good music quality.
However, installing ztdummy on FC4 causes moh to play very slow.
If I remove ztdummy, all works okay
Hi,
I encounter some problem with my Meetme application.
I am running on Asterisk 1.2.1 and kernel 2.6.9-22.0.1.EL
I got no problem when I comply with zaptel and ztdummy. However, it show me
the warning message something like this,
make: warning: Clock skew detected. Your build may
be
It looks like ztdummy isn't loaded. Compiled, but not loaded.
PaulH
- Original Message -
From: chan [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, December 09, 2005 1:14 PM
Subject: [Asterisk-Users] cant start the conference bride
Hi,
I encounter some problem
not sure if pop, the moderator of cofradia.org sells it, but you can post there and ask, they will help you out
best regardsOn 12/8/05, Claudio Canseco [EMAIL PROTECTED] wrote:
Hi list,
Im trying to build a system here at the office and i was thinking of buying this equipment
-- 1 TDM22B
-
On Thu, 2005-12-08 at 13:40 -0600, Chris Tooley wrote:
There's automatic snipe bidding, eBay wouldn't let us be an affiliate if
we did that.
I have to correct myself:
There's NO automatic snipe bidding, eBay wouldn't let us be an affiliate
if we did that.
Dear all
I’m
install the asteriskathome2.0,
According
to the asteriskathome handbook,
Download
the asteriskathome-h323-1.0.zip
When compile
the asterisk-0h323-0.6.5/asterisk-driver/chann-oh323.c
file
There’s
no asterisk/channel-pvt.h in the asterisk1.2
Have anybody
resolve?
Hi,Ever since upgrading to 1.2.0, Asterisk occasionally core dumps. I'm currently on 1.2.1 with the same problem.It crashes when an incoming call (zap) dials an extension. It will ring the extension short, then crash.
Here is the backtrace:#0 0x40187e06 in mallopt () from /lib/libc.so.6(gdb)
u need install pwlib and oh323 first,,then install asterisk-oh323...
hope this can help you..
On 12/9/05, paopaoerzhang(張志明=研發) [EMAIL PROTECTED] wrote:
Dear all
I'm install the asteriskathome2.0,
According to the asteriskathome handbook,
Download the asteriskathome-h323-1.0.zip
When
Dan,
I'd recommend reporting the problem along with a debug log and
traces from the router to: ooh323c-devel@lists.sourceforge.net
The developers have been quite responsive to problems I've had.
Thank you for tips.
I thing I must recompile the ooh323 module with 'make debug' then 'make
T/t/H/h and other options to Dial require Asterisk to stay in the RTP
stream.
George Pajari wrote:
We are trying to use Asterisk to set up a call between two SIP devices
and then step out of the path.
- all systems have public IP addresses (no firewalls, no NAT).
- sip.conf has
Douglas Garstang wrote:
Can someone tell me when SRV lookups are going to be fully supported in
Asterisk? I see we just had a new release, 1.2.1. Considering this lack of
functionality is a huge gaping hole for reliability, I would have thought 1.2.1
would have been a good time to implement
All,
I have an Asterisk system that sends PSTN calls to an OpenSER system to be
routed. I have a command like this in my extensions.conf:
exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
There's actually two OpenSER systems for redundancy. I'm trying to find a way
to have Asterisk attempt
Eric ManxPower Wieling wrote:
T/t/H/h and other options to Dial require Asterisk to stay in the RTP
stream.
Understood but already checked as not being the cause. Thanks for the
suggestion, though.
Here is our entire extensions.conf context:
[spa2100]
exten = _X.,1,NoOp(SIP Call from
Title: Can Asterisk accept and relay calls
I have a client looking for a cheap solution to relay calls from a remote site to their core voice switching gear.
The suggestion has eventuated to Asterisk being the box to accept the calls (from a Voice Carrier) via IP (and have a PRI) and then
Title: Can Asterisk accept and relay calls
The simple answer is yes, this can be done. Is there anyone
in Sydney? I dont know.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Skeeve
StevensSent: Thursday, December 08, 2005 8:51 PMTo:
asterisk-users@lists.digium.comSubject:
Hello, has anyone taken their cell phone number and ported it over to a
voip provider? If so, what voip provider and what was your experience?
Matt
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Title: Can Asterisk accept and relay calls
ACCA are in Sydney - if you need more info contact
me off the list.
PaulH
- Original Message -
From:
Kerry
Garrison
To: [EMAIL PROTECTED] ; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Friday, December
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