Hello David,
* Klaverstyn, David C [EMAIL PROTECTED] [09-05-07 09:40]:
We are currently using LDAPget 1.0rc6 with Asterisk 1.2.x. I see that
there is LDAPget 2.0rc1 for Asterisk 1.4.x. I was wondering if there
was something better. Are people using LDAPget or something else?
I have ported
Hi All,
Can anyone recommend any test kit that you can hook up your Pri/Bri
cards to without having actual ISDN in your office. For example
testing an * system before it goes to a clients office.
Thanks,
Gavin.
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We recently installed Asterisk 1.4.2
Tried to make calls using the Originate command (Asterisk Manager Interface)
All of the calls have zero billsec in the CDR.
Stumbled upon this:
http://bugs.digium.com/view.php?id=8680
so I guess the fix is not yet in 1.4.2.
Is this fixed in 1.4.3/1.4.4?
Hi Gavin,
A second Asterisk server replacing the provider (best way), or doing a loop
between two different ISDN ports on a same card (worst way) must help you.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Hi,
Everything was working fine on this 10 phone office, but for last few weeks
they are complaining that audio goes blank for a few seconds during the
conversation, and then comes back on. It goes blank for both parties.
What are the possible causes for this to happen?
--
Zeeshan A Zakaria
Hi Zeeshan,
Ethernet Network (or Switch) congestion ?
QoS not realy effective ?
Too high CPU load in Asterisk the server ?
Who knows...
You must check during a default.
Good kuck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
Jadrien Wauthier wrote:
Does anyone know how to adjust the volume of the ringing
application? I
have done a lot of internet searching and have not found much.
You cannot do this in Asterisk.
Some SIP phones might allow you to do so by setting an option on the
phone, but you would have to
Hi all,
I'm planning to deploy many Asterisk servers for remote sites connected
through IAX. Behind each server, there will be many sip clients
connected. A sip client from one site must be able to make calls for the
other sip clients connected to the other remote Asterisk servers. I've
On 09/05/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi Gavin,
A second Asterisk server replacing the provider (best way), or doing a loop
between two different ISDN ports on a same card (worst way) must help you.
Thanks for that. Will get a spare * box.
Best Regards,
Francois BERGERET,
That is not true regarding voice / fax detection with iaxmodem. If you are
running zaptel, then let it do the fax detection and have the iaxmodems
called from the fax context.
Craig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mail-lists
Sent:
Hi Calvis,
We have develop with MsCRM and Asterisk, if you still interested we are very
pleasure to help you.ç
Frank Bobbio
+34 932289310
www.icr.es
Barcelona Spain
[Asterisk-Users] Microsoft CRM Asterisk
calvis calvis at itechgroup.com
Any field return on this ?
Our last field trial of HPEC concluded we shouldn't use it at all, due to
audio clipping.
Is it now fixed ?
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I use DUNDi in this way, I have several remote sites and a MPLS
network connecting the sites. I have each sites asterisk box looking
at 2 DUNDi peers and those 2 central peers can query all sites. I
don't have a lot of phones or people moving between sites, but I did
not want to have to setup a
I use DUNDi in this way, I have several remote sites and a MPLS
network connecting the sites. I have each sites asterisk box looking
at 2 DUNDi peers and those 2 central peers can query all sites. I
don't have a lot of phones or people moving between sites, but I did
not want to have to setup a
I will be out of the office until Monday, May 14. Please contact OWD at
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Just the sake of curiosity, how many sites (or user) did you interconnect
using DUNDi ?
Regards
2007/5/9, Bruce Reeves [EMAIL PROTECTED]:
I use DUNDi in this way, I have several remote sites and a MPLS
network connecting the sites. I have each sites asterisk box looking
at 2 DUNDi peers and
Hi Ronaldo,
Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given
server can terminate to its peers. As a very simple example, if ServerA
houses extensions 500 through 599 and ServerB houses extensions 600 through
699, ServerA would advertise that it can terminate 5XX, and
8 maj 2007 kl. 15.40 skrev Joshua Colp:
Rohan Hathiwala wrote:
Hi,
I need asterisk to instruct the other side to send RTP to a
conference
server running on a different machine. The conference server does not
understand SIP so I cannot use the SIP REFER method.
I have another question.
I'm tying to use park and announce for call park on Asterisk 1.4.2 but
I'm having trouble getting it to work properly. This use to work with an
older version of Asterisk.
A telephone on the PSTN calls an IP phone. The IP phone is assigned
extension 3-8396. 3-8396 answers the call and
On 5/9/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
they are complaining that audio goes blank for a few seconds during the
conversation, and then comes back on. It goes blank for both parties.
I had this same problem happening every 8 minutes. It ended up being a
DSL issue at the DSLAM. You
There are nine sites, 10 servers. While it is not a huge deployment by
some standards, it was simplified with DUNDi.
On 5/9/07, Olivier [EMAIL PROTECTED] wrote:
Just the sake of curiosity, how many sites (or user) did you interconnect
using DUNDi ?
Regards
2007/5/9, Bruce Reeves [EMAIL
Alex,
Thanks for the linking to JR's article. That was my source for setting
up DUNDi also.
On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote:
Hi Ronaldo,
Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given
server can terminate to its peers. As a very simple example, if
Hi all,
I have an Asterisk server connected in a PABX (TELEDATA) by channel
Unicall (MFC/R2)..
I`m having problem when somebody call from PABX to Asterisk..
Eg: When somebody dial 1234, I received 113344 in the
Asterisk CLI...
If somebody can help me... or already saw
Damon Estep wrote:
http://www.asterisk.org/node/48317 does a nice job of explaining the
1.4 jitter buffer, however it raised a question in my mind.
In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the
UDP RTP packets renumbered on transmit, or is the original
We're actually getting two invites and schedules retransmit of both,
which is bad. One retransmit is stopped and the other one keeps
going, regardless of the ACKs that keep coming in. Needs to be fixed.
Believe I have fixed this in 1.4 svn, please test.
/O
- Patch
Index:
Hi;
Well, I understood now that Nortel has some digital
phones that can be used with astrisk, but the
question: what are the card models that should be
installed on Asterisk server? Digium? What these
models?
Regards
Bilal Ghayad
Hi all,
There is a problem with my dialplan. here is the dialplan:
exten= 123,1,Dial(SIP/U1,,Ttg)
exten= 123,2,Hangup
exten= h,1,AGI(onhangup.pl)
The problem is whenever U1 is called or calls someone, if U1 hangsup the
call then the h extension is NOT executed. but if the other person
Hello everybody,
I am receiving faxes and I don`t know how to receive, is there any posibility
to receive it on amail account?¿
in the console the message is this:
May 9 15:47:44 NOTICE[2618]: chan_zap.c:3703 zt_handle_dtmfup: Fax detected,
but no fax extension
-- SIP/101-0819b4f8
Hi,
Could you kindly send me that patch or give me the link to it. I am not
familiar with the bug tracker.
Regards,
Rohan Hathiwala.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E
Johansson
Sent: Wednesday, May 09, 2007 6:16 PM
To: Asterisk
Zeeshan Zakaria wrote:
Hi,
Everything was working fine on this 10 phone office, but for last few weeks
they are complaining that audio goes blank for a few seconds during the
conversation, and then comes back on. It goes blank for both parties.
What are the possible causes for this to happen?
Rizwan Hisham wrote:
exten = 123,1,Dial(SIP/U1,,Ttg)
exten = 123,2,AGI(onhangup.pl)
exten = 123,3,Hangup
exten = h,1,DeadAGI(onhangup.pl)
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Jonson Player wrote:
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database
with cdr. Well all I want is to receive incoming calls from pstn on
specified sip account (suppose 8000), and to initiate outgoing calls
from all my asterisk sip accounts through SPA3102 device.
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone calls
the Asterisk play a record (in gsm) and them the caller must choose a option
(1,2 or 3).
if he choose 1 it will redirect to 101 extension
if he choose 2 it will redirect to 102 extension
if he choose
Hello all.
We have an asterisk working perfectly but we need a sollution for the PA system.
Before Asterisk PBX we had an expensive analog PBX which plugged an
extension into an audio amplifier, and that was the PA system.
Now, the Asterisk server is quite far from the audio amplifier and it
Hello !
For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require
voltage on the line (although they don't use it to powerup and it just draws
a few mil amps)
As for PRI never tested, i would be interested to know how your test goes
Stelios S. Koroneos
Digital OPSiS -
I wonder if the your hardware is doing the actual DTMF detecting. What
hardware are you using? I'm using the TE205P and I believe that the
DTMF detection is being done in the software.
Remi
Steve Davies wrote:
On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote:
When dtmfmode is set to
I wonder if your hardware is doing the actual DTMF detecting. What
hardware are you using? I'm using the TE205P and I believe that the
DTMF detection is being done in the software in my case.
Remi
Steve Davies wrote:
On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote:
When dtmfmode is
Please try this,
exten= 123,1,Dial(SIP/U1,,Tt)
exten= 123,2,Hangup
exten= h,1,DEADAGI(onhangup.pl)
ok? ;)
Rafael Rodrigo M. Rosa.
www.megavoz.com.br http://www.megavoz.com.br/ (Voip e Telemarketing)
www.nsinet.com.br http://www.nsinet.com.br/ (Serviços Internet)
I will be out of the office until Monday, May 14. Please contact OWD at
800-337-3839 and ask for Client Services if you need immediate assistance.
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i suggest that you place it on a queue..
On 5/9/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone
calls the Asterisk play a record (in gsm) and them the caller must choose a
option (1,2 or 3).
if he choose
Hi,
My suggestion:
extensios.conf
exten = s,1,Answer()
exten = s,n,Wait(1)
exten = s,n,Read(MyVariable,TheNameOfSoudFile,1, , ,10)
exten = s,n,GotoIf($[${MyVariable}=1]?11)
exten = s,n,GotoIf($[${MyVariable}=2]?12)
exten = s,n,GotoIf($[${MyVariable}=3]?13)
exten = s,11,Dial(SIP/101,30,Ttm)
[Damon Estep]
I can see how bridging sip to sip via a zap channel would fix minor
jitter issues, since the zap timers are very accurate, however I cannot
see how this would correct out of order packets like a true jitter
buffer does (without the use of a jitter buffer on the sip-zap bridge).
SIP channel hang ups are progressively getting worse and I'm really
grasping at straws here trying to find out what the cause is. The
problem start, once a week or so the SIP phones couldn't communicate
with the server, though there was no error message on the server and
everything appeared fine
As usual, it is worth searching the WiKi for answers to this sort of question:
http://www.voip-info.org/wiki/view/Asterisk+Fax+to+email
This is not the only answer.
Regards,
Steve
On 5/9/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
Hello everybody,
I am receiving faxes and I don`t know
Ken,
I have similar problems every now and then on one of my asterisk boxes.
I'm also running CentOS4 on that box.
I've found that doing a sip reload when in that state results in
something along : Last reload not yet finished (can't remember the exact
wording)
We're using cisco 7960's
this is, just in case your expecting a volume of calls
exten = ,1,Goto(contexts,s,1)
[context]
exten = s,1,Answer()
exten = s,2,Background(support)
exten = 1,1,Goto(context1,s,1)
exten = 2,1,Goto(context2,s,1)
exten = 3,1,Goto(context3,s,1)
exten = 4,1,Goto(context4,s,1)
exten =
My suggestion:
[your incoming context]
#answer the phone
exten = s,1,Answer()
#playback recording but also accept extensions
exten = s,2,Background(your_gsm_recording)
#wait for caller to dial extension
exten = s,3,WaitExten(10)
#if they haven't hit an extension yet, play the message
That was in my list of things I've done, but failed to mention :). I
never have used DNS on this box, but for verification I removed DNS
servers and verified all addresses were IP's (which they were). There
is no DNS active on this box at all. There's also no freepbx, just
straight Asterisk.
whats the asterisk version your using?
On 5/10/07, Ken Williams [EMAIL PROTECTED] wrote:
SIP channel hang ups are progressively getting worse and I'm really
grasping at straws here trying to find out what the cause is. The problem
start, once a week or so the SIP phones couldn't communicate
Cheikhou DIAW wrote:
hi List,
i've been trying to get festival work on my 1.4.4 *box for the last 3days,
i've used the tutorial on this page
http://www.voip-info.org/wiki-Asterisk+Festival+installation
http://www.voip-info.org/wiki-Asterisk+Festival+installation
with exactly the same line in
Is there anyway to add additional volume gain
the console/dsp port?
I have used the mixer settings to set my volume on the soundcard to like
80 percent (I have even gone higher).
However I still need some additional volume when speaking to the
console/dsp.
With SOX I can do a -v X on files
Hello Folks,
I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I
have loaded the app_meet.so module in order to activate the MeetMe,
MeetMeCount and MeetMeAdmin applications. While I have been successful
in loading the app_meet.so module, I am experiencing an immediate
If you contact Digium tech support directly they will provide you with
the previous version of the echo canceler until the fix is made to the
current version.
Matthew Fredrickson
On May 9, 2007, at 7:27 AM, Olivier wrote:
Any field return on this ?
Our last field trial of HPEC concluded we
bilal ghayyad wrote:
Hi;
Well, I understood now that Nortel has some digital
phones that can be used with astrisk, but the
question: what are the card models that should be
installed on Asterisk server? Digium? What these
models?
None. There are no Nortel digital phones that work with
Started with 1.4.1, then 1.4.2, then 1.4.4, now the latest SVN (63478).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of franco
escalona
Sent: Wednesday, May 09, 2007 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Does anyone know if there is a known list of telemarketers?
Something like http://whocalled.us/ with an easier access?
We could all benefit if there was such a thing :-)
If there is enough interest, I could put up a database that everyone can
benefit from.
I just need some suggestions on:
(1)
Hi Folks,
Is there a way to find out the mobile/landline carrier name based on the
phone number?
For example, who is the mobile carrier for (415)2345678
I had heard about some query but just don't remember how/what?
Thanks in advance.
Ritesh
___
On 09/05/07, Stelios Koroneos [EMAIL PROTECTED] wrote:
Hello !
For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require
voltage on the line (although they don't use it to powerup and it just draws
a few mil amps)
As for PRI never tested, i would be interested to know how
I also get the mysterious SIP INVITE channels.
10.101.2.204 xxx 748e8b0a625 00102/0 unkn No Init: INVITE
And I also am running 1.4.4 on CentOS4. Is that a pattern or just
coincidence?
The other symptom you mention is this
...the SIP phones couldn't communicate with the
Regarding (2) - you can either provide a realtime query service supporting
web service interface which can be consumed using virtually any programming
language and it would be very easy to build an AGI script around it.
the second option would be to periodically update a flat file (csv) and
Try this:
http://puck.nether.net/npa-nxx/
*
Adam Moffett
Plexicomm, LLC
[EMAIL PROTECTED]
ph: 866-759-4678x104
*
Ritesh Agrawal wrote:
Hi Folks,
Is there a way to find out the mobile/landline carrier name based on
the
Hi Bilal -
Well, I understood now that Nortel has some digital
phones that can be used with astrisk, but the
question: what are the card models that should be
installed on Asterisk server? Digium? What these
models?
If you use the Citel Portico gateway, you don't need any telephony
card,
Can you connect existing Nortel system to Asterisk through fxs/fxo?
That way one could use existing infrastructure for few old phones and
Asterisk for new phones and all good things which come with it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi everyone, I have a user that needs a little extra volume on his
Polycom IP 601 phone set for all calls (beyond what the volume control
currently offers). Is there a provisioning setting for this anywhere?
(I'd like to avoid a separate amplifier between the phone and handset if
possible.)
Hi Ritesh -
Does anyone know if there is a known list of telemarketers?
Something like http://whocalled.us/ with an easier access?
We could all benefit if there was such a thing :-)
If there is enough interest, I could put up a database that everyone can
benefit from.
I just need some
On 5/8/07, Kevin Collins [EMAIL PROTECTED] wrote:
I modified chan_sip.c to turn on a dsp to do fax detect based on inband dtmf
being selected. And when reading rtp if 'f' character shows up vector to
fax extension
can i have your patched chan_sip.c ?
Kevin Collins
-Original
Go back to 1.2.x and see if it fixes the problem.
Ken Williams wrote:
Started with 1.4.1, then 1.4.2, then 1.4.4, now the latest SVN (63478).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of franco
escalona
Sent: Wednesday, May 09, 2007 11:02 AM
I mean that SIP phones cannot answer incoming calls or make outgoing
calls. When a call comes in on ZAP, it actually rings all the phones
like normal, but when you try to answer no one is there. In addition,
when you try to dial out you eventually get a message on the phones
saying unable to
Ritesh Agrawal wrote:
Does anyone know if there is a known list of telemarketers?
Something like http://whocalled.us/ with an easier access?
We could all benefit if there was such a thing :-)
If there is enough interest, I could put up a database that everyone
can benefit from.
I just need
Thanks Tim, good option.
The good thing with VPN is that two Asterisk servers would have no
exposure on public internet.
Tielin
[EMAIL PROTECTED] 05/08/07 1:45 AM
On 7 May 2007, at 19:51, Gordon Henderson wrote:
On Mon, 7 May 2007, Tielin Xu wrote:
Hi list:
Has anyone done to set up
-Original Message- From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: 08 May 2007 09:28 To: Asterisk Users Mailing List -
Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to
record CDR in DB Oracle On 7 May 2007, at 17:27, Florian
Vietnhi Phuvan wrote:
Hello Folks,
I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I
have loaded the app_meet.so module in order to activate the MeetMe,
MeetMeCount and MeetMeAdmin applications. While I have been successful
in loading the app_meet.so module, I am
This would not be valid, of course, for any number that was ported from
1 carrier to another.
Adam Moffett wrote:
Try this:
http://puck.nether.net/npa-nxx/
*
Adam Moffett
Plexicomm, LLC
[EMAIL PROTECTED]
ph: 866-759-4678x104
*
On 5/9/07, Adam Moffett [EMAIL PROTECTED] wrote:
Try this:
http://puck.nether.net/npa-nxx/
This probably goes without saying, but this data is, at best,
marginally useful due to LNP.
-erik
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Try this:
http://puck.nether.net/npa-nxx/
A better one is:
http://www.localcallingguide.com/lca_prefix.php
Note, however, that this will show the allocation of the NXX which may
no longer be the carrier handling the number if it has been ported to
another carrier. AFAIK there is no
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html
But it will be 3 PCI slots.
Thanks,
Gavin.
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asterisk-users
Noah Miller wrote:
Wow, that's a generous offer. I like the idea of a blacklist for
telemarketers. It's bound to be more effective than an RBL for
spammers! One thing to note: this may end up being a non-US database.
Here in the US, I've experienced great success with the
www.donotcall.gov
Hi,
Anybody using this Ericcson analog phone with Asterisk: Ericsson dialog
4187?
I was told some functionalities like CLID will only work with an Ericsson
PABX but other than that I would like to hear from anybody using this phone
on a FXS port.
Thanks,
Jose Limeres
On 5/9/07, Gavin Henry [EMAIL PROTECTED] wrote:
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html
But it will be 3 PCI slots.
You could do it in one slot with Digium's TDM2400P (you would actually
have to get
Thanks everyone for the responses, encouragement and offers to help.
I will get started on this shortly and circle back with you guys.
If someone has a starter list, it would help jump start the
efforts/motivation :-)
Ritesh
On 5/9/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Ritesh -
Does
Gavin - you should look at the Sangoma A4000X series cards, which only
occupy a single slot and come in PCI or PCI-X versions.
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: Wednesday, May 09, 2007 3:46 PM
To: Asterisk
I would look into one of these:
http://www.digium.com/en/products/hardware/analogcards.php
quote who=Gavin Henry
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html
But it will be 3 PCI slots.
--
And, did
Hello.
I have a customer that needs to downgrade the firmware on their SPA841 to
3.1.1(a). I can't seem to find the firmware file. Google turned up
3.1.2-something and Linksys is taking a while to get back to me.
Anyone happen to have that file lying around?
Thanks,
Nabeel
On May 9, 2007, at 3:45 PM, Gavin Henry wrote:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-
express-p-393.html
But it will be 3 PCI slots.
Just to clarify in case you didn't already realize it. It doesn't
actually *use* 3 PCI slots, it just occupies the physical space
Hello everybody,
Is there a possiblity to check in the dialplan whether a SIP user is
registred?
Something like :
exten = user1,1,GotoIf(isRegistred(user1)? context1, context2, 1)
Thanx,
Kalle
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[EMAIL PROTECTED] wrote:
Hello everybody,
Is there a possiblity to check in the dialplan whether a SIP user is
registred?
Something like :
exten = user1,1,GotoIf(isRegistred(user1)? context1, context2, 1)
Thanx,
Kalle
___
There will be a Twin Cities Asterisk Users Group meeting this Saturday, May
12th, at 11:30 'til about 1:30 at the Atacomm Corporate Offices at 7365
Kirkwood Court N., Suite 350, Maple Grove, Minnesota 55369.
Although there is no formal program scheduled, we'll chat about Asterisk
applications
Robert Augustyn wrote:
Can you connect existing Nortel system to Asterisk through fxs/fxo?
That way one could use existing infrastructure for few old phones and
Asterisk for new phones and all good things which come with it?
No. They are digital phones and use proprietary Nortel signalling.
Rizwan Hisham wrote:
Hi all,
There is a problem with my dialplan. here is the dialplan:
exten= 123,1,Dial(SIP/U1,,Ttg)
exten= 123,2,Hangup
exten= h,1,AGI(onhangup.pl)
The problem is whenever U1 is called or calls someone, if U1 hangsup
the call then the h extension is NOT executed.
Alvin Austin wrote:
Hi everyone, I have a user that needs a little extra volume on his
Polycom IP 601 phone set for all calls (beyond what the volume control
currently offers). Is there a provisioning setting for this anywhere?
(I'd like to avoid a separate amplifier between the phone and
Stephen,
I understand that these sets are digital but what about connecting Asterisk
fxs to Nortel fxo and keep sets connected to existing Nortel?
Would that work?
Robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stephen Bosch
Sent:
Its a PRI, no VoIP trunks, so no DSL. This happens only in the office, where
phones are connected through the same switch on which data flows for the
Internet traffic. But this started happening only few weeks ago. Is there
any way that I can check if its the switch or the router?
I have Grandstream and Aastra phones. It happens on both of them.
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On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote:
I understand that these sets are digital but what about connecting
Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel?
Yes you can do that; I have. No you don't want to; it doesn't work worth a
shit. You lose so
Robert Augustyn wrote:
Stephen,
I understand that these sets are digital but what about connecting Asterisk
fxs to Nortel fxo and keep sets connected to existing Nortel?
If you leave the Nortel PBX in the picture, I see no reason why that
wouldn't work.
-Stephen-
How about if both ServerA and ServerB houses extensions 500 throught
699. Such that users can dynamically register Server A or Server B.
Can we use DUNDi to implement such network?
On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote:
Hi Ronaldo,
Yes, you can use DUNDi for this. DUNDi simply
Andrew Kohlsmith wrote:
On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote:
I understand that these sets are digital but what about connecting
Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel?
Yes you can do that; I have. No you don't want to; it doesn't work
Hey, I'm hoping somebody knows the answer to this.
The script works fine on the old Trixbox 1.0 but have recently upgraded
(just testing in VMWare) to Trixbox 2.2
What happens is Trixbox will drop the call after I call the AGI command
in my dial plan.
I first of generate a call file to call
A small way to make little easy, I dont know it people are ok to that, try
integrating freepbx asterisk so you know what the sip configs should look
like when things are all well.
Things might stop working if there is a bug or change in configs.
--
Deepak
Ken Williams [EMAIL
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