hi abc def,
what type of voice codec that phone use. Maybe it
can't support.
I also have same problem my sip phone, when i
change the voice codec from
g729tog711 ulaw, then it work find.
also make sure wether your sip is behind the router
or not..
nat=never
or
nat=1
-
hi ,
i have downloaded the [EMAIL PROTECTED] software from the web ..but i have a little confusion about that ...either i wrote in blank cd as it is or some bootable media is required for it...as it is in zip format...BUT it is a .ISO file ...tell me ...what should i do...it will run automatically
Hi,
Use an application like the Nero etc to write .iso to a blank CD. Then you can use it on your spare computer to boot.
Remember you are going to lose all data on the reboot of the PC.
Keep kicking
Dan
On 26/01/06, Sohail Arham [EMAIL PROTECTED] wrote:
hi ,
i have downloaded the [EMAIL
Hi. Extension *.iso mean, that is image of original medium. U must write it
with burning sw as ISO image. Then u can access fs on your new medium.
Peter
-Original Message-
From: Sohail Arham [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 9:02 AM
To:
ahan...then it mean it doesnt need to uncompress it..juss write on cd by nero burning software...??
___
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yup... its a bootable image.. go ahead and just write it directly...
Dan
On 26/01/06, Sohail Arham [EMAIL PROTECTED] wrote:
ahan...then it mean it doesnt need to uncompress it..juss write on cd by nero burning software...?? ___
--Bandwidth and
Hi.
I am recording conferences taking place via the meetme application by
using the 'r' option.
When I start the conference I get the message in the CLI : Starting
recording of MeetMe Conference 8000 into file
meetme-conf-rec-8000-1138265171.201.wav.
No additional warnings or errors is
Damon Estep a écrit :
Can anyone point me to a reference or sample config for bypassing a
nailed up (point to point) t1 between two PBXs with asterisk and a
pair of t1 cards?
Right now I have 2 Nortel norstars connected to each other via a
leased line t1. I also have a solid 10mbps low
Jean-Michel,
You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING
A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I
MIGHT LOOK AT? Not WILL YOU DO IT FOR ME?
Your response to this post was un-informative and quite frankly it is the type
Bad day Damon? I think your comments are a little harsh towards someone who is an active and informed contributor to the list. Jean-Michel could have ignored you but he chose to share what he could. Maybe someone else will have the complete answer to your question.
On 1/26/06, Damon Estep [EMAIL
Damon,
I am not intimately familiar with what you are specifically trying
to achieve, *BUT*, if the two Norstars are essentially just
'interconnected' via teh T1 to provide either an EM Wink "type"
connection/private TDM bus between the two boxes so that extensions are
'bridged' between the
Hi I'm looking for a pinout for the above. Note this has what i'd call
RJ45 sockets (or someone smart can correct me). I need to plug into BT
(rj13?).
And, yes I've googled (glad I'm not chinese) and have tried the
suggested, just plug in a 6 connector rj11 and i didnt work atall.
On a
I have been using sipdiscount in sip mode (they are discontinuing their IAX2
connection) for a while. UK calls are free and its worked most of the time.
However, its not working this morning .-(
Chris
- Original Message -
From: RumaTech [EMAIL PROTECTED]
To: Asterisk Users Mailing
Actually, it is a quite appropriate
response to ANYONE that includes this type of comment in their reply
You probably need a
couple of T1 cards, and some paid consulting to get it working (I've never done
it myself but that's how I would do it if I was in a hurry)
Perhaps something
Hi all,
Ihave an IAX connection between two asterisk
servers and i'm looking for a way to cut down on the needed bandwidth. Both
voice and fax calls pass through the channel so it is currently configured to
use g.711.Could it be possible to select the codec based on the call's
prefix so
Damon Estep a écrit :
Jean-Michel,
You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING
A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I
MIGHT LOOK AT? Not WILL YOU DO IT FOR ME?
Yes, I think Asterisk can do what you are trying to
On Thu, Jan 26, 2006 at 09:42:36AM +0200, Mohamed Farid wrote:
Dear All :
I need to link my HQ to some Remote Sites - I need a Router which
supports VOIP , and VPN
Also the Router Should has 3 FXS ports and 1 FXO ...
The call should be routed from the Remote Site to the HQ through a VPN
Hello,
Can you provide a patch from your special branch for
asterisk-1.2.3 ?
can you post a how-to ?
Even these features won't be include in th main
branche a patch should be available.
Regards
harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
On 1/25/06, Douglas Garstang
[EMAIL PROTECTED]
On Thu, Jan 26, 2006 at 01:02:09PM +0500, Sohail Arham wrote:
hi ,
i have downloaded the [EMAIL PROTECTED] software from the web ..but i have a
little confusion about that ...either i wrote in blank cd as it is or some
bootable media is required for it...as it is in zip format...BUT it is a
Jean-Michel,
I agree with all of your comments, and would be willing to bet $100 that NO
AMOUNT OF GOOGLING will answer this question definitively.
After reviewing Adrian Carters very informative response regarding TDMoE I am
getting closer to what I need to know (now my googles include
I'm using an X100P Clone at home and i had not much trouble, remember
I'm just testing and learning a bit at home. I think if you hace to
implement it at office you'll have to spend a bit more.
2006/1/25, Joseph Tanner [EMAIL PROTECTED]:
Personally, I've had great success with an X101P (it's a
Why not try to purchase one of our GSM
Gateway at £60 and then you can route all the mobile calls through the GSM
Gateway?
http://cyber-telecom.net/store/product_info.php?products_id=29osCsid=4e787773c7c03212c43c51368d6ae387
Sam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Same situation.
Asterisk 1.2.1 ([EMAIL PROTECTED] 2.2) apparently doesn't have this problem.
Thanks
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Joseph Tanner
Sent: Wednesday, January 25, 2006 4:05 PM
To: Asterisk Users Mailing List -
On Wed, 2006-01-25 at 14:10 -0600, Kevin P. Fleming wrote:
Aaron Daniel wrote:
We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the
problem, everything started working. Doesn't seem like it's a bug in
1.2.1 :)
It is not. The bug was introduced during the 1.2.1-1.2.2
Hi Mimmus, and thanks for the quick reply.
You are welcome.
It is actually very good to hear that most of it works. The
difference in my project is that we'll keep the PSTN link on
the Alcatel, and use the asterisk only as a inter-site
trunking solution. The reason is that I have no
Hi
Does anyone know a good, scalable switchboard solution for asterisk?
I've been looking around and I've found a couple but I'm not sure yet...
Have anyone here used one in large environments? We need usable GUI
with the usual stuff like queues, transfer, meetme etc
roy
Move to PRI - it will be much more fun than working
with analog.
PaulH
- Original Message -
From:
Cisco - Kameko
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 6:17
PM
Subject: [Asterisk-Users] Digium
hardware
Hello,
I
hi
building a new setup, we want to try using sangoma cards. can these
be used as time sources the same way as TE410Ps?
thanks
roy
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In article 77758c190601240743o3ae310dbi28b2f79a93965776
@mail.gmail.com, [EMAIL PROTECTED] says...
I am not very satisfied with this, though. I want to use some
features (like Park) that apparently don't work well with reinvites.
Have any of the rest of you had any luck troubleshooting this
Thanks!
You are welcome.
Now the E1 is up, but still problems.
What I'm trying to do, is to let calls arrive to Asterisk
from the net, and using the Sangoma pass them to the PBX.
Is this possible?
Passing calls between different channels is the primary job of Asterisk, I
think!
You have to
Hello,
The 5 exchange lines I assume they are analogic. For them you will need 5
FXO ports. You can buy a TDM04B and a TDM01B (this will get you to the 5
FXO). Make sure you have 2 PCI slots available.
Now for the extensions you need
- IP Phones
or
- ATAs (if you want to reuse your analog
Hi,
is it possible pickup calls (with *8) between different channels (SIP and
IAX)?
Thanks
Mimmus
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Ronald Wiplinger wrote:
I tried to transfer a call, pickupcall and onetouch recording, but have
not got it to work!
You must uncomment the lines in feature.conf (remove the ; character
from the beggining).
--
Best regards,
Bartosz Piec
___
On 1/26/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,
Can you provide a patch from your special branch for
asterisk-1.2.3 ?
can you post a how-to ?
Even these features won't be include in th main
branche a patch should be available.
Harry -
There is a patch available against
I am using 0h323 on Asterisk CVS HEAD 19/07/2005. I am dialling a h323
gatekeeper. He can hear me, but I cannot hear him.
I have a suspicion that it could be the rtp traffic, since he said that
they need rtp traffic from ports 4500 - 65000. So in 0h323.conf i set
updstart and udpend,
Did you know that they switched over to a new set of servers? And they also
planning to switch off IAX very soon (as per their email notification to me on
the 13th of January)?
Von: RumaTech [EMAIL PROTECTED]
Datum: 2006/01/26 Do AM 07:35:49 CET
An: Asterisk Users Mailing List -
When you open your burning software there should be an
option to burn from an image. When it asks you for the
location tof the image point it to the .iso file that
you downloaded. After it is done burning the CD you
have a ready to go bootable CD. BE CAREFULL. Once you
put the CD into a machine it
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.
I assume with the mention of BT, you're in the UK.
Hi,
Try one of Venus 2804, 2808 or 2832 from Tainet corporation.
They support SIP or MGCP and they come with VPN.
http://www.tainet.net
Proceed to Product/VoIP/Venus
--
Regards,
Arek Bekiersz
Mohamed Farid wrote:
Dear All :
I need to link my HQ to some Remote Sites - I need a Router which
Chris Bagnall wrote:
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.
I assume with the mention of BT,
Short answer: Yes
Long answer: They use the zaptel drivers and are recognized as a
Zaptel device. You do have to load and configure the Sangoma wanpipe
drivers first, but in the end it'll function as a timing source just
like a Digium card
MATT---
On 1/26/06, Roy Sigurd Karlsbakk [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Damon Estep wrote:
Jean-Michel,
I agree with all of your comments, and would be willing to bet $100 that NO
AMOUNT OF GOOGLING will answer this question definitively.
I would almost be willing to take that bet... find your exact
configuration
I didn't right those products off and in fact use both on a regular
basis. For the price, both are pretty good. However, for a higher price
there are products on the market that _do_ handle echo cancellation in
a very solid fashion (eg, Mediatrix 1204 as one example) regardless of
the analog cable
That would not be a nailed up t1 - signaling at both ends would be via asterisk.
I was trying to determine if there is a way to configure asterisk to emulate a
ptp t1 passively (no signaling) - essentially providing the same type of end to
end circuit you would get if you ordered a point to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
http://www.voip-info.org/wiki-Asterisk+TDMoE
Damon Estep wrote:
That would not be a nailed up t1 - signaling at both ends would be via
asterisk.
I was trying to determine if there is a way to configure asterisk to emulate
a ptp t1 passively
TDMoE would allow a T1 like connection only over the local Ethernet segment,
since it is not an IP technology it can not be router across ip networks.
This would be useful to connect 2 asterisk boxes on the same Ethernet segment
(or with a crossover cable).
The advantage would be lower latency
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ok... lets get into the network setup... what about bridging a vlan
across your wireless network and sticking both asterisk on the same
segment? l2tp... (can a forgo the posting of the google links?) :)
Damon Estep wrote:
TDMoE would allow a T1
Damon Estep a écrit :
TDMoE would allow a T1 like connection only over the local Ethernet segment,
since it is not an IP technology it can not be router across ip networks.
You could use OpenVPN to create a virtual tap0 interface over IP, and
bridge that with your current ethX network.
Has anyone had any experience with the Linksys SPA-941 when it comes to
multiple line appearences?
This is what the 841 manual says: (maybe the 941 is different?)
The SPA-841 does not support multiple calls on the same Line key. The
corresponding Line key blinks quickly in red on any incoming
Lets put the TDMoE aside for a minute...
The same trunking could be achieved with SIP or IAX, could it not (with higher
latency)?
The rest of the question remains - is there a way to get asterisk to output,
bit for bit, on a t1 interface, the same data that is input on a remote
asterisk box
1000pps TDMoE plus vlan tagging, plus l2tp over 10mbps microwave?
I assume you have not tried this before, correct?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Thursday, January 26, 2006 6:47 AM
To: Asterisk Users
And in some (many) cases it will do so while sharing an interrupt with a
NIC and disk controller!
We run sangoma a104 cards in Dell SC1425 1U servers with great success
under heavy load.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Damon Estep a écrit :
Lets put the TDMoE aside for a minute...
The same trunking could be achieved with SIP or IAX, could it not (with higher
latency)?
The rest of the question remains - is there a way to get asterisk to output,
bit for bit, on a t1 interface, the same data that is input on
Sig Lange ha scritto:
I have successfully written FastAGI applications in python, and it
was a good experience.
Do you have some template code you can share ? or references to point us
to ?
___
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Damon,Unless I misunderstand what you are looking for, a P2P T1 would be handled by the kernel, not by asterisk. If you want to use digium cards, you would still need zaptel, or you could use a sangoma card on each end and their wanrouter drivers. Asterisk would obviously be involved in the SIP or
On 1/26/06, bails [EMAIL PROTECTED] wrote:
Chris Bagnall wrote:
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair I've got here have
What's in:
#include iax_additional.conf
#include iax_custom.conf
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
Damon Estep wrote:
I agree with all of your comments, and would be willing to bet $100 that NO
AMOUNT OF GOOGLING will answer this question definitively.
Um, if you google for pri_net pri_cpi and Asterisk, then I bet it will
return a response to your liking.
--
Cheers,
Matt Riddell
saw those, according to RAD they occupy 2mbps even when idle. about $750/each
for t1
From: [EMAIL PROTECTED] on behalf of Jean-Michel Hiver
Sent: Thu 1/26/2006 7:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] *
customer service sucks
as usual
I 100% agree. I havent been
able to complete a call ever. No response from customer service.
Whatever company can provide reliable
service, great support and a good selection of local numbers without charging
out the butt, will do very well IMO. Too
BJ Weschke wrote:
On 1/26/06, bails [EMAIL PROTECTED] wrote:
Chris Bagnall wrote:
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair
To
clarify: You have to write it as a DISK IMAGE. If you simply drag the ISO file
to your Nero project and write it, you will get a CD with a single file on it -
the ISO image - and not the CONTENTS of the ISO Image.
1. Run
Nero
2. In
the New Compilation dialog click Cancel
3.
Click
This has been an interesting discussion for me (except for the
sniping). The last post led me, out of curiosity, to this wiki entry:
http://www.voip-info.org/wiki-Asterisk+TDMoE
I was unaware of this feature, and it looks pretty good. I've been
pondering replacing some T1's by leveraging IP
You've clarified your requirements for me. Please indulge me - I really
want to understand - what are the application implications of this? In
other words, what system behavioral changes will your users experience
in the various scenarios (pure circuit emulation vs. relay via IAX or
Uhh..maybe you should ask
Jean-Michel for a refund.
Wait, you havent paid a dime for
this. Or Asterisk. Or most of the Asterisk add-ons.
I always see people getting mad at other
people for bad advice or bad answers to their
questions; people seem to forget that all this stuff is
Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable
(unless you encapsulate it somehow, I guess).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bill
Michaelson
Sent: 26 January 2006 14:58
To: asterisk-users@lists.digium.com
Cc: [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ross,
I was a little frustrated with Damon's initial reaction to the post as
well. However, we have moved past this ... This is actually turning out
to be quite an interesting thread, lets not get side-tract.
Regards,
Sean
Ross C wrote:
We are looking to replace our existing Legacy PBX with Asterisk. Our receptionist currently has a light display for a certain extension when someone is on a call. When she needs to transfer she simply hits that button.
Is it possible to use a snom360 + Sidecar to monitor 30 extensions and make
Right - so I will assume this makes it slightly more efficient in that
respect. And of course, any solution that uses multiple hops brings in
a raft of considerations for limiting interference by other data streams
- the essential QoS question.
Date: Thu, 26 Jan 2006 15:16:25 -
From:
Thanks Matt,
PRI signalling means that calls and answered and dialed (aka signalled) by
asterisk, the goal is to maintain the signalling between the two nortel boxes.
I have gathered that raw point to point circuit emulation is not possible on
asterisk...
I am aware of how to connect a PBX
gladly,
circuit emulation will;
1. eliminate the need to reconfigure the exisitng hardware.
2. improve the chances that fax and analog modem devices will still work.
3. NOT change any dialing patterns or extensons numbering.
there are other, but they are less significant
I've seen this discussion before. The conclusion was, it is possible to
route TDMoE through a VPN tunnel depending on the tunnel setup you are using
(bridge + tunnel for example) however the latency would make it useless.
TDMoE is designed for the same network. Unfortuanely I can't find a link for
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure
Damon Estep wrote:
saw those, according to RAD they occupy 2mbps even when idle. about $750/each
for t1
Are you basically looking to make a T1 repeater?
Or is there simply something that is removed from the signalling by
Asterisk that you want to maintain?
--
Cheers,
Matt Riddell
/attachments/20060126/d65904
6f/attachment-0001.htm
--
Message: 10
Date: Thu, 26 Jan 2006 14:55:22 +
From: bails [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TDM400 pinout
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Thanks a billion.
Outbound bluetooth dialling on the lines of
Dial(BLT/DevName/8005551212) worked for me.
Still trying out the inbound route. Before I created the [bluetooth]
context, it tried to reach the [default] context but then I began by
creating a new context [bluetooth] in
BTW, I did get clear bidirectional audio when I succeded in dialing
out...(with the channel = 3 in /etc/asterisk/bluetooth.conf) I have
Sony Ericsson T616 connected to a cheap commodity bluetooth USB dongle
that I bought ages ago from meritline.
On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
KalebIm atleast happy to hear that you get what you have paid for. I had not been able to get tru with international calls ever since the service was taken. I ad informed them and it takes ages to reply. They are asking be weird questions and any response again will be after a century (so to say).
Sorry not sure the mail was sent to the correct address:
--
Kind Regards
Etienne
---BeginMessage---
Hello all,
I was just wandering if it is possible to make Asterisk become a
replacement for an Ascend box and then utilise the unused channels to
make outgoing and/or incoming calls?
Possibly
She ain't cheap, but this'll work:
http://www.blackboxcanada.com/Catalog/Detail.aspx?cid=381mid=4291
It's TDMoIP so 2 T1 boxes tied together should work like this:
T1--TDMXX card--Asterisk--TDMXX card--Voice Mux--Broadband--Voice Mux--TDMXX
card --Asterisk
at about $7K Cdn it'd be worthwhile
Snom360 with Sidecar works perfectly. THe Cisco expnsion I
have yet to make work. I'll sell it to you if you want ( :-)
)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of c
waddySent: Thursday, January 26, 2006 10:31 AMTo:
Asterisk Users Mailing List -
I have contacted Digium and have received a quote of $7,000US to
implement what I will refer to as 'whisper mode'.
It will allow a person to speak to only one side of a bridged call. For
example, I am using ChanSpy to listen to an agent and what they are
hearing and saying. But I cannot tell the
Hi,
I've just
reinstalled Asterisk 1.2.3 on a fresh system and since I've noticed that the CDR
logging in MySQL (on a different computer) has stopped. I thought it
wasn't logging anything at all, but I realized after a bit of searching that
there were log files in
Hi! For reasons that I won't bore people with, I'd like to disable echo
cancellation on-the-fly, depending on which DID a call came in on. I've
seen things like spandsp disable EC for faxes, so I know it's possible.
Any idea where to start looking? (I assume I'll have to make a helper
Damon Estep wrote:
Thanks Matt,
PRI signalling means that calls and answered and dialed (aka signalled) by asterisk, the goal is to maintain the signalling between the two nortel boxes.
I have gathered that raw point to point circuit emulation is not possible on asterisk...
To connect
*** If anyone has a better way of doing this, please post to the list. I
hadn't seen anything on this list or in channel.c/chan_local.c - which
prompted this email ***
I'm not sure how many VoIP providers out there are using Asterisk as a
service platform like we do, but I thought I'd share
Arek,
Where can you get these?
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Arek Bekiersz
Sent: Thursday, January 26, 2006 7:50 AM
To: [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Thu, 2006-01-26 at 15:31 +, c waddy wrote:
We are looking to replace our existing Legacy PBX with Asterisk. Our
receptionist currently has a light display for a certain extension
when someone is on a call. When she needs to transfer she simply hits
that button.
Is it possible to use
On 1/26/06, Tomislav Parcina [EMAIL PROTECTED] wrote:
Hi Tomislav,
I am not very satisfied with this, though. I want to use some
features (like Park) that apparently don't work well with reinvites.
Have any of the rest of you had any luck troubleshooting this problem?
Your RTP stream
You can do a down and dirty test to see if it will work. You can
record the start of a fax tone into a file.
Then after you answer the channel play the file. The 'special tone' will
cancel all of the Ecs on the line.
Its dity but will work in a pinch.
-Original Message-
From:
What happened to the addmailbox script in version 1.2.3?
___
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currently, when using swift TTS engine with app_cepstral, generated audio is
streamed to the channel. This means that a call to ceptsral operates like
app_playback. I need the functionality of app_background. I'm thinking I
have two options... 1.) use system() to call swift engine, create a
Don't
need it. Add entries in voicemail.conf and mailbox is created on the
fly...
-Original Message-From: Tim Leeland
[mailto:[EMAIL PROTECTED]Sent: Thursday, January 26, 2006 10:47
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] addmailbox script
What
The script is silent!!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
LeelandSent: Thursday, January 26, 2006 12:47 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] addmailbox
script
What happened to the addmailbox
script in
OK, some update on this. It's not related to the Sipuras (actualy the
sipuras are very good at this, since they will re-ring your call). I
changed my setup to a mediatrix 1204 and I still have the problem.
Right now I'm looking at:
1. Changing the NIC.
2. Changing the machine asterisk is on.
I
-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160
Dear user,
the new snom 360 is able to use services from standard web servers.
Users can deploy customized client services with snom 360 and interact
with other
users via the keypad. The snom 360 will use HTTP protocol from
standard web
Hi there
Im having some echo problems on my snom 320 phones.
Anybody experience this before ? I dont have any issues with the sipura
841s I have though.
Any help is greatly appreciated.
Thanks !
Nora Lavelle
___
--Bandwidth
Available in the usual place.
ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0
This release includes minor spec changes, spandsp 0.0.2pre23, a new
Sangoma wanpipe RPM for use with the LSE kernel rpm and an AMP
installation document.
Best Regards,
--
Andrew McRory - President/CTO
Linux
I know this may be a backwards way but for several
reasons I have asterisk send all calls thru astcc.
With astcc you specify multiple routes with prioroty
settings. If it cant complete a call with one route it
will roll over and use the next one.
Regards,
Dovid
--- Cavanna, Richard [EMAIL
Title: Message
Hello all.
Anybody around that is utilizing the PauseQueueMember and UnpauseQueueMember
applications? Or even the AddQueueMember and RemoveQueueMember
applications? I'm trying to set these applications up to function in
relation to the agent number, rather than the extension
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