On Jul 25, 2006, at 12:52 PM, Mike wrote:
I didn't want to start a war either. It was simply an opinion that I
thought was worth expressing after reading all those GXP-2000 sucks
messages in the past.
It's still just an opinion, I am certainly not trying to build a
consensus.
Thanks for
Welcome to VoIP... Your operator needs to take care about
QoS when you are doing a download. Alternatively, there are some more-or-less
tricky and buggy tricks to stop downloads when you are talking; this needs to be
done on your IAD.
See for example http://www.voip-info.org/wiki-QoS.
CS
On 26 Jul 2006, at 02:00, Rich Adamson wrote:
Dan Austin wrote:
Stephen wrote:
If I connect two offices through an IPsec tunnel, what is the impact
on
latency, and does it noticeably affect calls?
That would depend a lot on the equipment that services the IPSEC
tunnel
endpoints.
Has
On 26 Jul 2006, at 03:04, Joseph Love wrote:
The issue which occurs is that the audio from the SIP client to the
IAX client will spend most of it's time sounded very robotic, and
garbled. It is possible, although very difficult to understand
someone who is on the SIP phone.
I have
Marco Mouta wrote:
By the way could any one tell me wich is the Bandwith with IP over
head for this codec. about 8kb/s?
Let's do some calculations on that:
g729a 20ms results in 20 bytes RTP payload in each packet, in order to traverse
the OSI model there's some headers that need to be
I tried sending this to asterisk-bsd a couple of days ago
I've been using Asterisk in various versions on FreeBSD for some time
now, but I've only just got to messing around with ACD.
I find I can't get any in-queue announcements to work, be they either
periodic or queue position announcements.
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.---BeginMessage---
Hi, i have installed asterisk and VICIDIAL call center and it's working fine couple days but when i reboot the computer there isthe problem.this is the asterisk -vvgc
Hi Friends, We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again.
hmm looks nicer than mine:
exten = *2002,1,System(asterisk -rx \
agent logoff Agent/ ${AGENTBYCALLERID_${CALLERID}})
exten = *2002,2,Playback(agent-loggedoff)
exten = *2002,3,Hangup
thx for your suggestion, i think i will integrate your solution
regard
KAI
Anthony Rodgers
Hi all !
I am currently planning a PBX asterisk installation in our new office.
We will slowly migrate from our old system to the new system, running
both systems paralel.
My question is now how to plan the extensions:
before we used to have only 2 digit extensions :
like 10, 70 etc. I guess
Watch your extensions don't conflict with local numbers - in Australia
1XXX numbers are valid!
PaulH
On Wed, 2006-07-26 at 10:32 +0200, Nik Engel wrote:
Hi all !
I am currently planning a PBX asterisk installation in our new office.
We will slowly migrate from our old system to the new
I have setup a SPA-3000 to forward all incoming PSTN calls to the asterisk
and for asterisk to use the SPA for outbound calls. This works fine, but is
there anyway to make the asterisk call the FXS port? So that I can call the
phone when needed and use the PSTN for calls if needed.
Thanks,
Dean.
Erik,
What a great and detailled explanation! Thank you very much!
Ps. If you know anything about legal issues asked abouta g729 please
post it here:)
Best regards,
Marco Mouta
On 7/26/06, Erik [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
By the way could any one tell me wich is the
Dear All,
I have bought a digium TE205p in order to move our E1 pri from a siemens
pbx to an asterisk server platform, I have already gathered the data needed
to configure the card but I am troubled by one thing that seems unclear
on all the documents I read.
The E1 is currently inserted in a
Pleasehow can I enable the 3way-conference on a sip
gateway like (addpac) by
dialing an extension example pressing
(*) since the gateway do not have this feature ,I want to make it on server level or if
you know the concept of how call-conference work .
Regards
Khaled,
This must be the fifth or sixth time you have posted this question. As
I'd written in my earlier mail to you, the reason why you're not getting
any reponse is maybe no one has what you want. Keep reposting the same
question won't get you the answers. It just annoys everyone.
Regards
If what you are asking for is a conference, you can use MeetMe and
transfer the participants to that MeetMe extension.
I you want it to be triggered by say the * sign then look at the
featuremap in features.conf. Using an AGI and redirect can do this for
you. Use the wiki @ www.voip-info.org for
Hi!
Does a ringing timer exist in asterisk to control ringing duration? If not,
is there a way to control ringing duration?
Thanks in advance for your help,
Michel
Message sent using UebiMiau 2.7.8
___
With pen in hand, Dean @ INKnBITs succussfully stormed bulwarks which others
armed with sword and excommunication have been repulsed, and said ...
I have setup a SPA-3000 to forward all incoming PSTN calls to the asterisk
and for asterisk to use the SPA for outbound calls. This works fine, but
If by ringing duration you mean how long a device will ring, then look
at options to Dial
If you mean how long the ring sounds to the callee look at
indications.conf
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zenone
Sent: Wednesday, July 26,
On 07/26/06 14:58 Phil Jordan said the following:
Before I get round to posting my configs for critique, is this a BSD
port issue? I see stuff around on the net re the BSD port, to the
no, it isn't a BSD port issue. many people run asterisk from ports with
ACDs without any problems. in
Mohamed A. Gombolaty wrote:
Dear All,
I have bought a digium TE205p in order to move our E1 pri from a
siemens pbx to an asterisk server platform, I have already gathered
the data needed to configure the card but I am troubled by one thing
that seems unclear on all the documents I read.
On 7/26/06, Paul Hales [EMAIL PROTECTED] wrote:
Watch your extensions don't conflict with local numbers - in Australia
1XXX numbers are valid!
And similarly emergency services 3-digit numbers, 112, 999, 911 etc.
In fact I would avoid numbers that are even similr to this. 1112 could
easily be
Hi Ismir.
It tries to change of slot its TDM400P, I find that it must resolv.
Good luckJosué
2006/7/26, ismir saljic [EMAIL PROTECTED]:
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls.
Great rates starting at 1¢/min.
-- Mensagem encaminhada --From:ismir saljic
Dear Steve,
Yes I did mean a csu/dsu I will try your suggestion and update the results.
Thx
MAG
Steve Totaro wrote:
Mohamed A. Gombolaty wrote:
Dear All,
I have bought a digium TE205p in order to move our E1 pri from a
siemens pbx to an asterisk server platform, I have already
Ps. If you know anything about legal issues asked abouta g729 please
post it here:)
if you are briding g.729, without transcode, and you will NOT stay in
mediapath (canreinvite=yes), you don't need g.729 licence
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL
Alex Robar wrote:
Hi all,
I have a Sangoma A200 card with hardware echo cancellation. The card has
12 ports (10 of which are active; All FXO). Twice on this particular
card I've seen all ports simply stop receiving incoming calls. There is
no other indication of this, however. I am able to
Here is what I
do...
Exten=777,1,AgentCallbackLogin()
Yup, thats it! use
your agent id and password, and then enter your dialable number. I say dialable
number because you can basically dial any phone number. We have agents that call
a toll free number and login to their home phones,
But my question was, is it possible to free the channel if it rings too
long?
Michel
Message sent using UebiMiau 2.7.8
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asterisk-users
I have a wip-300 linksys phone, tdm2402e and a number of uniden phones.
When I call wip-300 to any internal phone it sounds just fine.
When I call outside on the tdm2402e board it is not clear. the other
person does not hear
anything odd but I hear drops in the audio.
When I call out with a
Zenone wrote:
But my question was, is it possible to free the channel if it rings too
long?
Yes. show application dial in the Asterisk CLI will show you where
the timeout goes on the Dial line.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Is there a table available, which tells me if a zip code, city and area
code matches?
For now I did it with google, type each info in and found out if it
matches, but it would be easier if there is a table available.
bye
Ronald
___
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Bill Gibbs wrote:
Randomly, and this is very hard to debug because it happens so quickly
on outbound calls I get a one way screech, it's a steady tone that's
very loud. The remote end cannot hear it. You can hear the person
talking through the tone. I can't describe it but it's bad enough
On Wednesday 26 July 2006 00:03, marek cervenka wrote:
i'm reading a lot docs about asterisk realtime
but i cannot understand how works sip realtime static
i need NAT/qualify for SIP. this is not possible with dynamic realtime
i want
- save data to sql
- asterisk -rx reload to read config
Hi,
We purchase the database with zip codes, latitude, longitude, are codes and
all for our zip lookup AGI.
If you need something simple take a look at
http://www.census.gov/geo/www/gazetteer/places2k.html
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
So would this be the remote end echo can freaking out or the Polycom on
the caller side?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Wednesday, July 26, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial
Ronald Wiplinger schrieb:
Is there a table available, which tells me if a zip code, city and area
code matches?
I doubt, that such a table does exist.
Imho you will have to look for individual tables for each country.
For Germany, look at:
http://w3logistics.com
Roger.
- Message d'origine
De: Eric ManxPower Wieling [EMAIL PROTECTED]
A: Zenone [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 12:54
Zenone wrote:
gt; But my question
IP Phone - Asterisk - PSTN.
This would be the Echo Canceler on the Asterisk/Zap - PSTN interface.
Bill Gibbs wrote:
So would this be the remote end echo can freaking out or the Polycom on
the caller side?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Quoting Ronald Wiplinger [EMAIL PROTECTED]:
Is there a table available, which tells me if a zip code, city and area
code matches?
For now I did it with google, type each info in and found out if it
matches, but it would be easier if there is a table available.
If you subscribe to the LERG,
Ok, in my case it would be my Cisco 3660 since that's what talks to the
PRI. It talks sip to my Asterisk box.
Thanks!
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Wednesday, July 26, 2006 9:34 AM
To: Asterisk Users
Then none of this applies.
Bill Gibbs wrote:
Ok, in my case it would be my Cisco 3660 since that's what talks to the
PRI. It talks sip to my Asterisk box.
Thanks!
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent:
Hi,
Is it possible to have a CSTA support for asterisk...
If possible how to configure it
sanchal
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Will do. I'm already starting to like the Polycom 501, but then again I`m
not a typical technically incompetent end-user with no desire to learn
anything new.
I can see a big learning curve for the customers who go from 5 cascading
phone lines on PSTN phones to a VoIP PBX. I believe the
Hi
I have the following setup:
SPA3000 (at home) -- Asterisk1 server (at home) --- Asterisk2 server
(at work).
The reason the SPA3000 isn't connected directly to Asterisk server 2
is because the SPA3000 can't register to more than one SIP account at
a time, plus it was more fun that way :)
I get round this bug by replacing:
exten = X,1,Dial(sip/blah)
with:
exten = X,1,Answer
exten = X,n,Dial(sip/blah)
It means the call is in an answered state before it starts ringing but
it doesn't seem to cause any major problems.
Mike
Martin Schrott - Thinking-Systems wrote:
Hi all,
I
I just bought a brand new TDM400P but it came with all 4
cards, not realising I only needed 2.
I now have FS: 2 x
Asterisk X100M (red) daughterboard cards brand new.
Email me your best offer and your location for a confirmed delivered price. (Im
in New York 10021).
Payment must be
Here is my setup, this is just a test lab til I figure out how to do this
Both machines are on a lan, no routers, firewalls etc between
BoxA
192.168.1.192
2XXX extensions
BoxA iax.conf
[boxb-peer]
username=boxa-user
type=peer
trunk=yes
secret=mypassword
host=192.168.1.139
[boxb-user]
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
change ${EXTEN:1} to ${EXTEN}
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Your cutting the leading dialed number from each box
exten =
_2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _2XXX,2,Congestion
should be
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN},30,r)
exten = _2XXX,2,Congestion
Bails
Tim P wrote:
Here is my
while that did seem to change the error message it still doesn't ring the other phone
here is the error message:
-- Executing Dial(SIP/2001-781d, IAX2/boxb-user:[EMAIL PROTECTED]/1001|30|r) in new stack
-- Called boxb-user:[EMAIL PROTECTED]/1001
-- Hungup 'IAX2/boxb-peer-1'
== Everyone is
IAX2/boxb-user:mypassword2 at 192.168.1.139/1001|30|r) in new stack
it's dialing extension 1001, not 2001
ah, i seetry this (dirty, but it should work)
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/2${EXTEN:1},30,r)
___
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Great that fixed it, i can now call to the 2xxx extensions from my 1xxx extensions
awsome, thanks so much!
On 7/26/06, whois wes [EMAIL PROTECTED] wrote:
IAX2/boxb-user:mypassword2 at 192.168.1.139/1001|30|r) in new stackit's dialing extension 1001, not 2001ah, i seetry this (dirty, but it
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension,
if the extension hangs up first, the next line of the dialplan gets
executed,
if the pstn hangs up first, shows exited non-zero on ZAP/6-1 and
the next
2006/7/26, Mike Diehl [EMAIL PROTECTED]:
We have ISDN phones that have a Message Light that we don't want to break.Hi Mike,How will these phones be connected ?Regards
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Hi, We are using "Asterisk" in our office and using "XLite" as softphone and your service for making calls to USA. When I am using SIP, Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, it is telling that "I am sorry. That is not a valid extension. Please
2006/7/26, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Hi, Is it possible to have a CSTA support for asterisk... If possible how to configure it
sanchalHiFor curiosity, why would you like Asterisk to support CSTA ?Do you have any legacy applications or devices needing it ?Regards
If you need something more than that, it will be difficult. A zip code can
serve multiple NPA-NXX's
and an NPA-NXX can be in multiple zip codes.
Don't forget that number portability significantly muddied the waters, and
VoIP has created an environment where there's no longer any need for a
Hi,Which ATA supporting T.38 would you recommend (for reliability) ?Has anyone experienced this one ?http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102
Regards
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asterisk-users
On 7/25/06, William Piper [EMAIL PROTECTED] wrote:
On 7/25/06, Randy Paries [EMAIL PROTECTED] wrote:
Hello,
I just got my Asterisk up and running, and everything is great
What i can not seem to find is a doc that describes any of the user
commands
Like is there things like, end message or
On Mon, 24 Jul 2006, Douglas Garstang wrote:
Not for our users. We held focus groups, and the Polycom's won in terms of
ease-of-use over all the other phones investigated.
Which other phones did you investigate specifically?
Our users found the polycom menus cumbersome, with commonly used
James Fromm wrote:
Yeah, we tried that. Tried every combination of variables in sip.conf.
Only solution that works is removing the requirement for a secret.
Faris Raouf wrote:
One thing to try is setting type=peer instead of type=friend. I'm a
bit dazed and confused at the moment, but
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 26, 2006 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Just bought a Polycom 501 - I feel
likemyGXP-2000was better...
On Mon, 24 Jul
Hi Tim,
Thanks for the suggestion. Unfortunately the jitter buffer does not
seem to be the culprit. With it on or off, the issue still occurs.
I'm still playing with some testing of codecs to see if it's codec-
related, as per Rich Adamson's suggestions, and will continue
discussion of
Hi
I have been asked if it possible to connect a SE F250M to Asterisk. I
have never used one of these devices before but from what I have
gathered they need a FXO interface. As the Asterisk box is hosted
remotely would it possible to use a Sipura 3000 to provide the FXO
interface and successfully
And similarly emergency services 3-digit numbers, 112, 999, 911 etc.
In fact I would avoid numbers that are even similr to this. 1112 could
easily be mis-dialled as 112.
sure good point thank you, will adopt that to german emergency numbers,
nevertheless 4 digit
gives me most flexibility ?
I didn't test it with a Sipura, but a TDM400. You can check this page
for configuration codes for the F251M.
http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In
Spanish). If the SPA-3000 supports detecting polarity reversals,
you'll need them.
Julian.
On 7/26/06, Jon Farmer
Hi all !
I am planing to set up around 20 SIP Phones which will be purchased in
one bunch, I am more or
less free of choice.
I wonder if anyone knows sip phones which allow configuration upon
login. The following scenario:
User logs into any phone and the settings of the phone are always the
Thank you for replying, Dinesh.
Right, if all the cryptic comment I found just refers to the Zaptel
stuff, that isn't a problem. Thank you for the clarification. To
business:
My queue timeout is 240 seconds. My periodic announcement interval is
30 seconds. Queue position announcements
On Jul 26, 2006, at 8:58 AM, Phil Jordan wrote:
I tried sending this to asterisk-bsd a couple of days ago
I've been using Asterisk in various versions on FreeBSD for some time
now, but I've only just got to messing around with ACD.
I find I can't get any in-queue announcements to work, be
verify property of dev/zap; if your asterisk running in non-root mode;
change /dev/zap chown into asterisk non-root user.
Regards.
Jon Schøpzinsky wrote:
Hello list
We are having some strange problems.
When we setup trunking between two of our servers, the connection only uses
trunking one
Julian J. M. wrote:
I didn't test it with a Sipura, but a TDM400. You can check this page
for configuration codes for the F251M.
http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In
Spanish). If the SPA-3000 supports detecting polarity reversals,
you'll need them.
Thanks for
Hi Group!Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users.We got an small ISP and we have the project to give telephony (for now) to our users between
- Original Message -
From: Carlos Alberto Bernat Orozco
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wed, 26
Jul 2006 16:49:03 -0300
Subject: [asterisk-users] Developing VoIP with
Asterisk
Hi Group!
Greetings and salutations.
Still I'm concern about my problem
Hello all,
I was having some trouble earlier with Asterisk mis-hearing my
extensions (this is when dialing into a DID from PSTN). For instance,
if I dialed 1234 it might hear 122334.
I was using Asterisk 1.2.7 and SIP routing at the time, and I upgraded
to Asterisk 1.2.9.1 and SIP and things
- Original Message -
From: Olivier
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 26 Jul 2006 14:18:29 -0300
Subject: [Asterisk-Users] Which ATA to test
T.38 ? What about Linksys 3102
Hi,
Which ATA supporting
- Original Message -
From: Crazy Boy
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Wed, 26 Jul 2006 13:58:39 -0300
Subject: [asterisk-users] SIP is not working
sometimes. IAX is working fine. Why?
Hi,
We are using Asterisk in our office and using XLite as
- Original Message -
From: Jean-Yves Avenard
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 26 Jul 2006 11:19:34 -0300
Subject: [asterisk-users] Message waiting
question...
Hi
Hola!
I have the following
Dear All,
I have a strange problem in recieving calls on the pri the zaptel
is green and everything seems very well, but when a call comes I can see
the call along with the caller ID but then I get this strange message which
make the call hungup:
error msg: 'zap-in' from '0109687348' does not
Dear Steve,
The line has worked like charm, but now I am facing a new problem with
recieving the call, I have sent another mail with this issue.
Thank you very much for your support
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Steve,
Yes I did mean a csu/dsu I will try your suggestion and update
I really like the IP60x phones. Have started using the IP430, so far
after 20 or so they are fine.
But the IP30x and 50x I refuse to use.
The aastra 480i is also good.
The 9133i has promise.
I do not like the snoms - any.
Grandstream are so so
Budgetone is not bad for the price, but not
- Original Message -
From: Mohamed A. Gombolaty
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Wed, 26 Jul 2006 18:40:07 -0300
Subject: [asterisk-users] Strange Error when
calling
Dear All,
Greetings.
I have a strange problem in recieving calls on the pri the
This looks like a dialplan problem - do you have a match for
0109687348 in the zap-in context in your dialplan?
A.
On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote:
Dear All,
I have a strange problem in recieving calls on the pri the zaptel
is green and everything seems very well,
I have a customer who is HOH (Hard of Hearing) and needs the volume on his
handset set to the maximum volume level. Currently he has to manually set
the volume to the max on every phone call that he makes which is a pain.
How do I set the volume to the max and have the phone remember that volume
Need to configure the volume persist parameter in the config file, I
do not think it can be set on the phone directly.
On Jul 26, 2006, at 5:04 PM, calvis wrote:
I have a customer who is HOH (Hard of Hearing) and needs the volume
on his
handset set to the maximum volume level. Currently
find this line in your sip.cfg file. 1= remember last setting,
0=return to default
volume voice.volume.persist.handset=1
voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/
At 05:04 PM 7/26/2006, you wrote:
I have a customer who is HOH (Hard of Hearing) and needs the volume on
Manrique Feoli wrote:
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension,
if the extension hangs up first, the next line of the dialplan gets
executed,
if the pstn hangs up first, shows exited non-zero on
Yes, on a Zap FXO channel, when you can hear ringing, the timeout is
counting down, even if the remote party hasn't answered yet.
Zenone wrote:
- Message d'origine
De: Eric ManxPower Wieling [EMAIL PROTECTED]
A: Zenone [EMAIL PROTECTED], Asterisk Users Mailing List -
Hi,
Does any one knows some thing about this issue?
I'll appreciate any comments!
Telles
Rodrigo P. Telles wrote:
Hi,
I'd like to know if some one knows how to make Asterisk record a call after
xfer (not bxfer).
I tried some ways but it doesn't work at all.
extensions.conf example:
On 7/26/06, Randy Paries [EMAIL PROTECTED] wrote:
On 7/25/06, William Piper [EMAIL PROTECTED] wrote:
On 7/25/06, Randy Paries [EMAIL PROTECTED] wrote: Hello, I just got my Asterisk up and running, and everything is great
What i can not seem to find is a doc that describes any of the user
thanks for your worthy advise Andres, that in deed does the trick.
I had actually thought about that solution, but then I'll have to
evaluate all calls again at hangup ( h ) to see how to handle their
end,
That in my case wasnt all that nice given I need different types of
When I looked several months ago, the only Sipura that supported T.38
was the SPA-2100. I haven't searched in a while, but I think it is
still true. We go directly from a Cisco gateway to the SPA-2100 and it
works great. It is the only ATA that we've seen that works right.
Joshua Colp
Hello,
Does anybody have experience with the Quad T1/E1 PRI cards in an
HP DL380? Just a yes it works fine or a never again is enough :-)
Edwin
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog:
On Thu, Jul 27, 2006 at 10:06:35AM +1000, Edwin Groothuis wrote:
Hello,
Does anybody have experience with the Quad T1/E1 PRI cards in an
HP DL380? Just a yes it works fine or a never again is enough :-)
I've had a couple of Digium cards in a DL360 working fine, no problems
at all.
Steve
--
Sangoma's response so far is that it is a known issue. They have heard of it from a few customers, but find the issue extremely difficult to produce (and have not yet been able to replicate the problem in their labs, even on their long-term tests). The suggestion (for now) is to update to the
Might be easier to share the directory over WebDAV. Only need to have one port open on the work firewall (if in place) to allow access and can also run it over SSL.-brandonOn 7/26/06,
Joshua Colp [EMAIL PROTECTED] wrote:
- Original Message -From: Jean-Yves Avenard[mailto:[EMAIL
On Thu, 2006-07-27 at 10:06 +1000, Edwin Groothuis wrote:
Hello,
Does anybody have experience with the Quad T1/E1 PRI cards in an
HP DL380? Just a yes it works fine or a never again is enough :-)
It works fine with a TE210P card. I did turn off the hyperthreading.
Regards,
Patrick
Hi.
Thank you so much for answering. I guess I couldn't get a better
qualified answer !
On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote:
Anything is possible, it's just to what extreme do you want to go to make it
happen. Right now we have no way of transporting arbitrary information (like
Anyoneaware of a way to turn off the call waiting beep via tftp for
cisco 7960's? Disabling this through the call menu doesn't appear to
work.This would be for sip firmware
Thanks
Cory J
AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY
14225++voice - 716.630.1555
Peder @ NetworkOblivion wrote:
When I looked several months ago, the only Sipura that supported T.38
was the SPA-2100. I haven't searched in a while, but I think it is
still true. We go directly from a Cisco gateway to the SPA-2100 and
it works great. It is the only ATA that we've seen that
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