At the moment when one of our users wants to transfer a call, they press
the transfer button on the phone, enter the extension and away they go.
Is there any way to do this via the AMI or dialplan ? I want them to
push a button on the screen rather than using the phone itself.
Julian
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good unless the
incoming call comes from isdn, which means via the avm fritz!card. In
this case
Hello Christopher,
an Asterisk callback agent can be anywhere, even on a POTS number. He will
have to register with a number that can reach him as far as Asterisk is
concerned. I don't see the scenario you are proposing as particularly
difficult to implement in Asterisk.
Hope this helps
l.
Grandstream have acknowledged that there is a problem with 1.1.0.13 on
later phones (MAC's 00:0B:82:09:xx:xx I assume) and have advised me to
wait for the next firmware release. So anyone with later phones (MAC's
00:0B:82:09:xx:xx), do not upgrade to 1.1.0.13.
On Wed, 14 Jun 2006 [EMAIL
Hi!
I've added member to a queue like this, from queues.conf:
member = SIP/[EMAIL PROTECTED]
It works OK. But, after restaring I see in show queue that
Members:
SIP/[EMAIL PROTECTED] (Invalid) ...
What does it mean? Why is it Invalid? BTW, reload command fixes it, so
the member
Hi,
I have been playing around with the latest release of asttapi and I have
found the 'hangup' problem already reported to the list here
http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html
Apparently hangup should be done by making use of UserEvent commands. So I
have
More than 128ms?
128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but
definitely more than 16ms.
No, 128ms = 1024 taps
Like what sangoma offers.
Ding, Ding, Ding, Ding!
Okay, to be complete in my answers:
No I do not get more than 128ms delay caused by European routing (I
I will try your suggestion and I will let you know. Thank you On 6/18/06, Philippe Lindheimer [EMAIL PROTECTED]
wrote:How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show
Hi all,
We have a SIP trunk with * and even when there are calls in progress
sip show inuse always shows 0 calls in progress. I have outgoinglimit
and incominglimit limit set and have also tried call-limit. sip show
inuse works fine with SIP handsets though very frustrating.
Hi Steve,
Thank you for your answers. First of all span 3 is not a satellite link
and no echo occurs when I connect this line to another pbx with HW EC
feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I
have to do something to enable EC for this card ?
Idris
-Original
On 6/19/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hi Steve,
Thank you for your answers. First of all span 3 is not a satellite link
and no echo occurs when I connect this line to another pbx with HW EC
feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I
have to do something
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good unless the
incoming call comes from isdn,
My Telco is bringing a T1 line to my company. It will be delivered via
Copper. From my research on the net and in this group, I've found out that I
have the following options:
· OPTION 1 - Ensure that the My PBX Equipment (CPE) provides a T1
interface.
· OPTION 2 - Convert the T1 into 24
Hi Dakota, I think that you would have to opt to the first option, why T1 must be digital, where you he must have a TE110P inyour Asterisk. Of preference it opts to ISDN, therefore total it is supported in asterisk and much more easy of if programming. I wait to have helped
GreetingsJosué
Hi,
Im using the Read command the read a DTMF tone.
In this read command I play a voice-file.
But now when I press one off they keys of my telephone the
voice-file will stop playing a the program go the next priority.
Is it possible to play the voice-file until the right DTMF
tone
Correct,
And no, I am not passing H.This was identified as a bug in the
chan_agents code.
On 6/17/06, Wes Baehr [EMAIL PROTECTED] wrote:
Create a context for your queue and put a '*' extension to redirect them
back to the main menu (or wherever)
Also, you're not passing option 'H' to
On 6/18/06, unplug [EMAIL PROTECTED] wrote:
Hi,
Does asterisk support mutl-port binding? Say beside setting the
port 5060 in sip.conf, I want to use another port, say 6060. How can
I set to use more than one port. Is it possible?
unplug
Not possible in Asterisk, but you should be able to
Greetings All,
The Ultra 5 will take Solaris 10 no problem, however RAM will be an
issue. Be sure that there is at least 128MB of RAM on these units or
Solaris 10 will tend to chug. The SparcStation, from everything I've
read, is not supported under Solaris 10. You can, however, get older
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Monday, June 19, 2006 1:48 pm
Subject: Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm
fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
Hello everyone,
I have Asterisk SVN-trunk-r7498
What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine.
bp
On 6/19/06, Eric Bishop [EMAIL PROTECTED] wrote:
Hi all,We have a SIP trunk with * and even when there are calls in progress sip show inuse always shows 0 calls in progress. I have outgoinglimit and
I have tried it with 1.2.7.1 and 1.2.9.1. Same issue with both and only on the SIP trunk, not on endpoints.On 6/19/06,
William Piper [EMAIL PROTECTED] wrote:
What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine.
bp
On 6/19/06, Eric Bishop
[EMAIL
I liked the ringer that read the phone
number too, but a couple months ago, I did a firmware upgrade, and that ringer
option went away Do you have the latest firmware?? I upgraded because
of a problem with my phone losing registration, which is now fixed, but I lost
that really cool
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Monday, June 19, 2006 1:48 pm
Subject: Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm
fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
Hello
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
At the moment when one of our users wants to transfer a call, they press
the transfer button on the phone, enter the
If you know which channel you want to transfer, then one way is to use
the Redirect AMI action
(http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Actio
n+Redirect).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent:
On Sunday 18 June 2006 13:38, Brian Capouch wrote:
The hardware is pretty crappy.
I never had any real trouble with the QuickNet PhoneJack PCI cards (I have
three, one I blew out the SLIC because I hooked it up to POTS and someone
rang me), but then again I haven't touched them in probably two
Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a
x86_64 box (I guess nobody is using x86_64 platform or is able to fix this
themselves?)
First of all when bristuff is downloaded and compile is started it appears
that the bristuff Makefiles are badly broken.
The
I am looking for wireless SIP phones that will
also receive a text message.
Has anyone phone such a phone?
Thanks,
jerry
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I have an issue where someone will park a call, and then it will ring
back to them, but because the caller-id looks like a regular inbound
call, they don't know how to answer the call (these are the
receptionists).
I've tried to make an extention that I can transfer to that will set
the
Given that the NSLU2 can't do trunking, do you think that a PIII
733Mhz, 128MB RAM will do?
Thanks,
Daniel
On Jun 15, 2006, at 4:15 AM, Tim Panton wrote:
On 15 Jun 2006, at 02:59, Daniel Salama wrote:
Does anyone know how many simultaneous calls can a WRTG54GS
handle? Assuming SIP
Check
http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.html
On Sun, 18 Jun 2006, John Millican wrote:
Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
I have a data T-1 available to me to do some testing of a new asterisk
systemthat I am putting together. Do I just leave this T routed through
my cisco router and plug in the asterisk system through a network card
or do I need to get a T-1 card and use that? I looked on the voip-info
wiki and it
Warren,
My suggestion for testing would be just use ethernet hand off to the asterisk
from the Cisco. You could bypass the Cisco but then you would need a T-1 card
for the asterisk box and they are not cheap. I believe there are valid
arguments for both choices though and ultimately should be
You don't need a T1 card for a data T1. Just run it through your Cisco box send it over to your NIC on the asterisk box.
bp
On 6/19/06, Warren [EMAIL PROTECTED] wrote:
I have a data T-1 available to me to do some testing of a new asterisksystemthat I am putting together.Do I just leave this T
I have a data T-1 available to me to do some testing of a new asterisk
systemthat I am putting together. Do I just leave this T routed through
my cisco router and plug in the asterisk system through a network card
or do I need to get a T-1 card and use that? I looked on the voip-info
wiki
John,
Thanks for the quick reply. I do intend to get a T-1 card anyway.
Would it be the same card for a data T-1 as for a voice T-1 just with
different setup?
W
John Millican wrote:
Warren,
My suggestion for testing would be just use ethernet hand off to the asterisk
from the Cisco. You
Depends what you want to do!
Do you want to do VoIP over that T1 to a provider or IP telephones?
Do you want to hook up to the PSTN through that T1 as 24 voice channels,
through a T1 card on your asterisk?
If you want to use the T1 as 24 voice channels, the Telco is going to
have to re-provision
On Mon, Jun 19, 2006 at 03:41:30PM +0200, Remco Barendse wrote:
Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a
x86_64 box (I guess nobody is using x86_64 platform or is able to fix this
themselves?)
First of all when bristuff is downloaded and compile is started it
John,
Well I am certainly not an expert on this. I am using an SPA-3000 and I
have not experienced this. I did have to go to inband on the fxo channel
as rfc8322 did not work for ivr's when using Asterisk. I think you said
you were using a linksys or sipura product for you fxo?? If that is the
Warren,
Yes. The setup is based on what type of signaling the telco is giving you.
John
On Monday June 19 2006 10:32 am, Warren wrote:
John,
Thanks for the quick reply. I do intend to get a T-1 card anyway.
Would it be the same card for a data T-1 as for a voice T-1 just with
different
Doug,
thanks for the help. I am using uLAW and inband every where. I have tried
using 2833 and it did not appear to make any difference, better or worse.
this is why I was thinking that if I could increase the minimum required time
for a tone that it night help, I am just not sure where the best
I have looked thru the users lists, google and voip-info and did not find an answer. I am using asterisk (latest SVN as of three weeks ago). I am using real time. I have a problem that when a user enters an invalid meetme extension the meetme says invalid room and dumps the call. Here is what I
Steve,
I want to end up with a system that will let me send and receive voice
calls. I guess what I want to do depends on the best way to do that.
Can I do more than 23 (decent sounding) voice calls on a data T-1 with
someone else handling the final part of the call to the copper for me?
If so
Is anyone using the HDLC facility in Zaptel to bring a data T1 into an
Asterisk system? I know this was available in kernel 2.4.19--is anyone
using it in kernel 2.6.x?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
I have just finished implementing an Asterisk system for my place of
business (first one), and after three days of flawless usage, Asterisk
seems to have crashed. I wasn't running with '-g', so I don't have a
core dump. Here's the sequence of events leading up to the crash:
1. call comes in on
If you get it figured out, please post details on the wiki. I tried
about a year ago. I think I was close but I didn't have enough time to
pursue it. Looks to be trivial with Sangoma though I haven't tried that
either.
Thanks,
Steve Totaro
-Original Message-
From: Michael Welter
I suspect that the majority of the advice that you are going to get
would be to upgrade to the latest version of asterisk, as so many
changes and bug fixes have been made since the 1.07 release.
Julian.
Mark W. Stoddard wrote:
I have just finished implementing an Asterisk system for my
Steve, that happened to me too. I downloaded the public release (not beta) and it was included. I noticed that the new firmware includes a different ringer. I guess they decided we didn't need that ringer.
Do you update off of their system, or do you have your own tftp server?
On 6/19/06, Steve
All,
Slightly off topic.
Polycom released their SIP software version 1.6.6 for their phones recently. I
was under the impression that this release fixed a previous limitation where
the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk.
I have configured a phone
Who Carez? wrote:
Hey all,
I am running a Asterisk 1.2.9.1 with Sangoma A101 card, newest firmware,
configured with a help from Sangoma Tech Support, running fine. It is
connected to a PRI circuit split from Cisco MC 3810, which in turn is
connected to a Converged T from CTC Communications.
Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1.
Remember, you have 1544Kbs of bandwidth.
g279=8Kbs per call
uLaw=64Kbs per call
Just do the math.
bp
On 6/19/06, Warren [EMAIL
John,
You said you were using a PAP2.. what is the FXO interface at the (far)
asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
*
Remember to add the RTP, UDP and IP overheads.
And
then just do the math.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of William
PiperSent: 19 June 2006 17:12To: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: Re:
Thanks for the tip. No idea why I missed this.
Off the top of your head, does this support attended xfer, or is it a
blind xfer facility ?
Julian.
Moises Silva wrote:
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Double check to make sure you are actually running 1.6.6. I have it
working with 14 extensions right now with no problems...
Sean
Douglas Garstang wrote:
All,
Slightly off topic.
Polycom released their SIP software version 1.6.6 for their
- Denis Shaposhnikov [EMAIL PROTECTED] wrote:
What does it mean? Why is it Invalid? BTW, reload command fixes it,
so
the member receives queue calls.
I've just reviewed the code and this should be working properly... please do a
'set debug 3' and enable the 'debug' channel in logger.conf
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to offer any suggestions.
The only requirement is that 'echocancel=yes' is
- Douglas Garstang [EMAIL PROTECTED] wrote:
Polycom released their SIP software version 1.6.6 for their phones
recently. I was under the impression that this release fixed a
previous limitation where the phones would only watch 7 buddies, ie
send 7 sip subscriptions to Asterisk. I have
Uys, Steve Underwood
I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I
get the far and local end unblocked but as soon as I try to make a call I
get dialing and then protocol failure..
Do you guys know if there are any issues with sangoma and unicall? Anybody
has an
The latest version of Asterisk also includes a Page command so that
you can use that instead of an AGI script.
On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I suspect that the majority of the advice that you are going to get
would be to upgrade to the latest version of asterisk, as
Anyone on the list good with Linksys
PAP2NA configuration, I am looking to take my atas and emulate the
operation of a pots phone line as close as I can get. One thing I need to
change is the fast busy tone I get when someone hangs up on the call.
We are
Hi,
Thanks to those who hinted on the SIP/H323/Skinny capabilities of
asterisk ... I am starting to like this app! :D
Now, I successfully managed to bridge SIP to H323 (i don't have skinny
phones here). Just a question: Is it possible to have Asterisk just
as a signalling proxy? i have a flat
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten = 2,8,Monitor(wav,${TIMESTAMP})
exten =
All:
I tested echo test by dialing *43 under Asterisk configured by FreePbx
by using x-lite softphone. I could not figure out how the call is routed
to context from-internal. In sip_additional.conf, I have three
extensions defined as 2826, 2800 and 2801, which all are defined context
as
Hey everyone,
I recently bought an Act-Tel G2DS telephony gateway (the
web interface says it's model # is GS though.) Has anyone else on this
list used one of these? It has one FXO and one FXS port. I have an account
for it set up in sip.conf on my Asterisk box and it apparently logs
All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools.Thanks-John
___
--Bandwidth and Colocation provided by
So let's assume I am going to use G.729A. I am looking at using
Polycom IP601 phones which support G729A directly, so the only licenses
I believe I would need are for the calls going to voicemail or in the
menu system at once - realistically that number never exceeds 5
simultaneous, since the
Thanks Noah for the help, but... no go :-/
From: Noah Miller
ONE: You should answer an incoming zap line before doing anything with it,
so do this:
exten = s,1,Answer
exten = s,2,Dial(Zap/2/014XX)
When I try this, instead of using the Zap/2 interface to ring the other
number,
arp in the shell
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any network tools.
Thanks-John
as long as they are in the same network segment as the asterisk server
you can use arp
man arp
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any
After all the overhead, for uLaw you would need
about 90kbps (give or take) and for G.729, you would need about 32kbps (give or
take). Therefore, you would have the following:
uLaw= about 17 calls
g729= about 48 calls
I am trying to start a voip service in my local
area and sometimes
Doug,
The interface that i dial to is at my Service provider and am not sure what
they are using. I dial out of my * box to a service provider number which is
answerd by an asterisk box that I have at another location on a high speed
cable connection, that box then dials the numberI ultimately
[EMAIL PROTECTED] root]# arp -an
? (172.16.8.1) at 00:04:C1:21:CC:C0 [ether] on eth0
? (172.16.8.53) at 00:04:F2:01:FA:94 [ether] on eth0
? (172.16.8.48) at 00:04:F2:01:FA:D8 [ether] on eth0
? (172.16.8.62) at 00:04:F2:01:FB:65 [ether] on eth0
? (172.16.8.60) at 00:04:F2:01:FB:20 [ether] on eth0
Kevin P. Fleming wrote:
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to offer any suggestions.
The only requirement
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any network tools.
Thanks-John
From
your Asterisk console:
tcpdump -i eth0 -e | grep -A1 your target phone's IP
address
Then:
Make a
call on your target phone.
Disclaimer: not tested
-Original Message-From: mojowrkn
[mailto:[EMAIL PROTECTED]Sent: Monday, June 19, 2006 11:21
AMTo:
On Mon, Jun 19, 2006 at 10:21:16AM -0700, mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a phone
on my system?? I can get the ip's just fine but dont seem to be able to pull
mac addresses from any network tools.
Is it in your LAN?
if so, arp(8) is your
If they are on the same network you can do the following:
arp -a | grep $IPADDRESS |awk '{print $4}'
you may need to adjust awk(ed) position due to you distro.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Casey Boone
Sent:
I cannot help you with the problem, I can only tell you it works for me (on a
Debian system)
I wonder what the florz patch is though.
I never used it, but I hear some ppl about it all the time.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
an example might be
IP=10.0.0.213
MAC=`arp | grep $IP | awk {'print $3'}`
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any network tools.
If your phones are connected to a Cisco switch, depending your your IOS
level you can possibly use the show mac-address-table command. Which would
show you not only the mac-address for all the devices attached to the
switch, but what port they are hanging off of.
Hope this helps.
T.
If your T1 is currently configured for connecting you to the Internet,
then your Asterisk just becomes a client on your network, and can
terminate calls to Internet based providers by SIP or IAX. No reason
for a T1 card or connection to the Asterisk. I don't have enough
experience to say who may
I found a message on this list, that provided a recommendation to use
195.140.132.34, which I think is a non-afflilated someone that just happens to
be providing tested firmwares. I couldn't get the default to work...
What server do you use? What firmware do you have? I've got a GS100...
I honestly do not see the big deal about using g729. It is a one-time
fee and you would only need to buy as many licenses as you have people
in ivr or voicemail if you have g729 phones. For a business this is
not a major expense. You are talking about spending $100-$200 (max $480
for all 48
If youre going to have to open
ports on your firewall for SIP anyway, then why not put the server on the
inside? That being said, I dont know if youd need to punch holes
for the phones being trusted and the server on the outside..
Personally I dont like the ideas of
having a server
I would say its only profitable if
youre getting ONE T1 instead of two??
From: Gabriel Afana
[mailto:[EMAIL PROTECTED]
Sent: Monday, June 19, 2006 1:34
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How
to use a data T-1?
I have a very, very simple Asterisk setup in my house. I have a
Sipura 3000 with a PSTN line connected and one analog phone connected.
The [incoming] context looks like this:
exten = s,1,Dial(SIP/50,23,r)
exten = s,2,VoiceMail([EMAIL PROTECTED])
exten = s,3,Playback(vm-goodbye)
exten =
On 6/19/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Kevin P. Fleming wrote:
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to
On Mon, Jun 19, 2006 at 01:52:39PM -0400, Alexander Lopez wrote:
If they are on the same network you can do the following:
arp -a | grep $IPADDRESS |awk '{print $4}'
grep before awk?
arp -n | awk '/^IPADDRESS / {print $3}'
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile
again it is required to change KSRC=/usr/src/linux/ to
KSRC=/usr/src/linux-2.6/
I wonder why neither florz nor kapejod fixes these problems (several
modules do not compile).
I will not try running bristuff anymore without florz but from
With recent versions of *, you can increase the detection time in
zapata.conf with the toneduration variable.
The default setting is:
toneduration=100
We had the same problem and solved it by increasing this to 200.
You can also increase the threshold volume for detection of DTMF by
I am running an IP601 on my desk and it is only monitoring up to 8.
If I add more, it drops the oldest and adds the new one.
running 1.6.6.0036
On Jun 19, 2006, at 11:40 AM, Kevin P. Fleming wrote:
- Douglas Garstang [EMAIL PROTECTED] wrote:
Polycom released their SIP software
Well that is certainly all good news. The last hardware question I
would then have is: What do you do for Echo Cancellation with this type
of setup? Everyone keeps saying that the software EC basically sucks to
put it bluntly. Is there some sort of hardware to do EC that can be
used here?
W
19 jun 2006 kl. 19.02 skrev Cesc:
Hi,
Thanks to those who hinted on the SIP/H323/Skinny capabilities of
asterisk ... I am starting to like this app! :D
Now, I successfully managed to bridge SIP to H323 (i don't have skinny
phones here). Just a question: Is it possible to have Asterisk just
Using the Background command, you will be able to play the voicemail
while still being allowed to enter digits.
exten = s,1,Wait(2)
exten = 108,2,Background(voicemail/default/108/unavail)
exten = s,1,Dial(SIP/50,23,r)
exten = s,2,Background(/voicemail/default/50/unavail) ;or whatever the
I'm looking for somehting like the standard house hold linksys/dlink router.
Basically it needs to have at least 1x100mbit port, wireless G capabilitys
and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2
which appears to do what i want. Anybody use this? Does it
True, there have been many fixes since then. I would at least consider
upgrading Asterisk+zaptel to the latest 1.0x which I think is 1.09.
If you want to try troubleshoot it first I would watch my memory usage over
the next couple days for memory leaks. If you find your using more and more
Shaun,
I believe that there are 2 models of the WRT54GP2 as there was/is with the
PAP2's one that is set for VONAGE and one that is not, typically referred to
as the WRT54GP2-NA
John M
On Monday June 19 2006 3:37 pm, Shaun wrote:
I'm looking for somehting like the standard house hold
Matt,
Thank you very much!
I am currently running 1.2.7.1 but will be upgrading to 1.2.9.1 this week. I
will try toneduration=200 first and let you/list know how well it worked.
I read in zapata.conf.sample where it says:
How long generated tones (DTMF and MF) will be played on the channel
Not quite sure. Audiocodes gives a dialtone when the number is called from PSTN. After few seconds I see the SIP invite to the Asterisk box. Asterisk responds with SIP 404 .Thanks,Lal
On 6/12/06, Erick Perez [EMAIL PROTECTED] wrote:
So is the problem with your audiocodes or with the asterisk
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