I'm running pretty much the same config in Belgium.
Here's what I use:
zaptel.conf:
span=1,1,0,ccs,hdb3 # no CRC4 used here
bchan=1-15,16-31
dchan=16
zapata.conf:
[trunkgroups]trunkgroup = 1,16spanmap = 1,1,1
17 maj 2006 kl. 12.13 skrev Samuel Tardieu:
I am trying to register my Asterisk server to a SIP server which
doesn't accept an Expires: field smaller than 1800 seconds and
indicates it correctly with a Min-Expires: in an error response when
Asterisk tries to use its default of 120 seconds.
Is
I follow the advice of Alasdair, it was happening because of the multiple kernel panics. I have installed it again, and now it's working properly!
Thanks a lot for your help. I'll change also all default passwords for security reasons.
BR,
//Laura
From: Steve Jones [EMAIL PROTECTED]Subject: RE:
On 5/17/06, Hadley Rich [EMAIL PROTECTED] wrote:
They do, but it isn't released yet. Put B410P into google and you will get a
couple of hits. Digium's marketing page says it is available and the
distributor I use had one on show the other day so they can't be too far
away.
Aside from being
The BT guy should check LD 73 block LPTI and prompt AFF.
If it is crc then you need crc4 as well.
Best regards
Hans
Steve Totaro schrieb:
Andy Kirby wrote:
I am new to the group but have searched the doc's FAQ's etc before
posting here.
We are attempting tie our asterisk server/service
On Thursday 18 May 2006 18:35, stoffell wrote:
Aside from being available.. What driver does it use?
Will it be needing bristuff ? (that wouldn't work I guess)
Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
And thus, making bristuff obsolete? (wich means, BRI users
AMP dialplan is full of garbage and perpaphs is not fully 1.2 compatible but
it is anyway an Asterisk, working dialplan!
I already tried to copy config files and Asterisk starts without warnings:
gradually I will clean out them from fax, queues, devices, ring groups,
weather reports, etc
quote who=Sharon Lim
hi there,
i am wondering can meetme.conf able to support diffferent context. Cause
currently, it has [rooms] context. ]
is it possible to have same conference number with different context?
thanks
Try it and see ;-)
--
Kind Regards,
Gavin Henry.
Open Source. Open
Hi Christopher,
i know the place, in fact i've been reading a lot before post here.
The problem is that even if there are a lot of good documents,
personallly i can't see answered my doubts, and this is the reason i
wrote.
If you can be a bit more explicit and give me some light over my
Hi Oliver,
i understand you use astlinux, even if the version nunber shows the
product is quite new. If you have to decide between astlinux and
[EMAIL PROTECTED], thinking in use the pbx for basic thinks like MOH, IVR,
an advanced dialplan, 1 FXO, 3FXS, 3 SIP and no much more, which one
would
I have a TE110P connected in euroisdn as pri-cpe.
When I dial out from a sip phone to a number over the pri, I get an error
Unable to set channel 1 (index 0) to linear mode
On the destination phone, I only get a terrible noise when answering the call.
There doesn't seem to be a speech path...
Hi Michael,
your document is very good. In fact this was one of the first i read.
I googled looking for soekris and asterisk and you appreared.
Anyway, your document do not cover the same setup i have. You point
the problem of the digium cards don't using it, while i have or i
think i require
Andres wrote:
Hi Klaus,
The response to a CANCEL should be a 487 Request Terminated, not a
200 OK. Maybe your innovaphone Server is to blame.
Hi Andres.
No. The reply to the CANCEL is a 200 Ok. The reply to the cancelled
INVITE is a 487.
regards
klaus
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote:
Otherwise the Diva server cards
are a good option (extensive, but come highly recomended from most that
I hear). Good luck and happy hunting.
Ouch, you weren't joking. 1453 Euro!
--
Cheers
Wayne
Hello every body.
I have this PCI card : DM/V1200-4E1
spec in this site:
http://www.intel.com/network/csp/products/3967web.htm
Can i use it with Asterisk, is it compatible ?
Thank you in advance.
AMP dialplan is full of garbage and perpaphs is not fully 1.2
compatible but it is anyway an Asterisk, working dialplan!
As example:
May 17 18:35:40 WARNING[8625] app_db.c: This application has been
deprecated, please use the ${DB(family/key)} function instead.
May 17 18:35:40 WARNING[8625]
On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote:
are a good option (extensive, but come highly recomended from most
that
I hear). Good luck and happy hunting.
Ouch, you weren't joking. 1453 Euro!
But worth every penny, imo. I have a few servers running Eicon Diva
Server V-4BRI cards and
Hello,
I am trying to use a trunk SIP between my Asterisk server and a Biling
prepaid server.
Problem: I would like to disable the authentication trunk what is the
command for that request.
In my log server I have: proxy authentification required
Regards
Rabii
On 19:16, Thu 18 May 06, Avi Miller wrote:
On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote:
are a good option (extensive, but come highly recomended from most
that
I hear). Good luck and happy hunting.
Ouch, you weren't joking. 1453 Euro!
But worth every penny, imo. I have a few
Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to
Kevin P. Fleming wrote:
Klaus Darilion wrote:
Shouldn't there be some error indication if Asterisk discards a response?
Probably, although it's not clear here that Asterisk actually discarded
anything. Without seeing the entire dialog, there's no way to be sure
whether there were multiple
* Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
R is not a valid Dial option.
Sure about that? My Asterisk installation lists it as a valid option.
asterisk*CLI show application Dial
[...]
R- indicate ringing to the calling party when the called
party indicates
[...]
r
Thanks for the advice. indications.conf is now existent and
Asterisk is reloaded but the problem still persists.
Reloaded? Peraphs restarted is better...
DV
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Hi there,
I've been playing with a SNOM 360 and 190 trying to get them talk to each
other using g722 with 16 kHz. However all I see in the SIP log codec
negotiation is g722/8000 which makes me believe that this is only a 8
kHz link (and that's what it sounds like).
Anyone every managed to
Hi there,
this is now the second time I've seen an issue like this with 1.2.7.1,
the first time it was a DNS hickup, today its some Internet congestion:
When one (!) or more register statements in sip.conf fail the entire
Asterisk becomes very unresponsive and does not accept registrations
* Sebastian Kayser [EMAIL PROTECTED] wrote:
are there any caveats regarding ringing indication with Asterisk?
I have got an asterisk installation with a quadBRI driven by BRIstuff.
Internal phones are various snoms (320 / 360) connected via SIP and
Idefisk softphones connected via IAX2.
Hello every body.
I have this PCI card : DM/V1200-4E1
spec in this site:
http://www.intel.com/network/csp/products/3967web.htm
Can i use it with Asterisk, is it compatible ?
Thank you in advance.
Hii all
I bought te110p card.
I configured zaptel.com ;
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16 and zapata.conf is;
switchtype=euroisdn
signalling=pri_net
group=1
context=nortel
dchannel = 16
channel = 1-15,17-31 I receving the following error
on my gentoo;
May 18 13:48:44
* Sebastian Kayser [EMAIL PROTECTED] wrote:
* Sebastian Kayser [EMAIL PROTECTED] wrote:
are there any caveats regarding ringing indication with Asterisk?
PSTN -- 3 x BRI -- POTS (NEC) -- 3 x BRI -- Asterisk
^ ^
|
Hello Gentelmen,
I am in china, just ordereda tdm21B card (2 fxs and 1 fxo), still waiting for its delivery.
Whether anybody already tried this kind of card here in china, will you please tell me how it works, any issue?
esp.
1) caller id
2) dtmf
3) busy tone
4) hang up
Thanksa lot!
Thank you for your quick response. I have successfully implemented
Intercom (Dialling within my office LAN) using Asterisk. To implement this,
I am using X-Lite Softphone.
Now, I want to make calls to US using VoIP Asterisk. I think that there is
no need of any external hardware to
Hi,I'm trying to start with Asterisk, but I could not put 2 softphones to talk.The asterisk server rejects the connections always when I dial.May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from
192.168.0.106What is necessary to put it to work?There is no need to configure external
* Mimmus [EMAIL PROTECTED] wrote:
Thanks for the advice. indications.conf is now existent and
Asterisk is reloaded but the problem still persists.
Reloaded? Peraphs restarted is better...
The reload messages informed about the re-reading of indications.conf.
However, even restart doesn't
Emre BALCI wrote:
snip
May 18 13:48:44 ERROR[8755]: Channel 24 is reserved
for D-channel.
did you change the jumper setting to E1 as per
http://www.digium.com/en/docs/TE110P/te110p_config.php?
Looks like the card thinks its a T1 card.
Leo
___
Yes I changed jumper settings but I receiving
following error;
May 18 14:43:43 WARNING[8481]: Ignoring canpark
May 18 14:43:43 WARNING[8481]: Ignoring dchannel
May 18 14:43:43 WARNING[8481]: Unable to specify
channel 1: No such device or ad
dress
May 18 14:43:43 ERROR[8481]: Unable to open
Does your AAH box have a static IP address or is it a DHCP client? Run ifconfig to check the IP address on the card. On 5/17/06, Brian McCarey
[EMAIL PROTECTED] wrote:
Hi all,
Weuse AAH to run our office telecoms
registered with two Sipgate accounts.
Today, Sipgate appeared to have had
Looks like you're on the right track with this.
I have just diffed the 'i' version with the 'p' version and found the same
line as you found below.
I've changed the 'peercallstate' line in the 'p' version with the
'ourcallstate' one and compiled.
I tested the outbound and inbound dialling over
Jonathan Gonzalez wrote:
I'm thinking buy only de SBC and look for another chassis where all
equipment fit fine. I say you don't cover, or this i think, because i
need to use FXO-PBX-FXS/SIP and you don't use this setup.
Don't attempt to use a Digium or other FXO and FXS card on a soekris
Before I naff around changing the network card, as anyone got any useful thoughts. I think when I pulled out the cable, the card went on the blink..!
Run ifconfig as root and see what that tells you about your eth connections.-- Justin BiggsOwner, Biggs Computer Consulting
[EMAIL
Thanks for the feedback, but the route that that I'm finding doesn't
work is:
Asterisk - SPA3000 - ZAP/BRI - Asterisk - DISA
The problem appears to be on outbound calls from the SPA3000 where the
second dial tone seems to stop audio transmission, changing the DTMF
method make no difference.
I'm trying to start with Asterisk, but I could not put 2 softphones to
talk. The asterisk server rejects the connections always when I dial.
May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106
What is necessary to put it to work?
There is no need to configure external
Hi. I have a aplication for web, when u press on
the link, the application log into an asterisk(user, password), and call to one
extension(ex 201). How can I do to that call go to 201, if busy, go to 202, and
so on? I want to implement in a call center.
Best Regards
Thanks
Ever
On Thursday 18 May 2006 08:07, Darrick Hartman wrote:
Don't attempt to use a Digium or other FXO and FXS card on a soekris
board, especially if you plan on using a full-featured distribution like
CentOS (which is what AAH uses). See more below...
I read below, but you don't explain why not to
Hi,
Is there anyway I can make a softkey light on a sip phone (aastra
9133i) light up when the agent is logged into a queue? Even if I have
to do it via some call in a dialplan. I guess the question is more...
what command do I need to send to a sip phone to turn a light on a
softkey on/off?
I'm trying to start with Asterisk, but I could not put 2
softphones to talk.
The asterisk server rejects the connections always when I dial.
May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from
192.168.0.106
Try puting a
permit=0.0.0.0/0.0.0.0
In the sip.conf for your two phones.
stoffell wrote:
Aside from being available.. What driver does it use?
Will it be needing bristuff ? (that wouldn't work I guess)
The Digium B410P will use the mISDN stack and chan_misdn for Asterisk.
Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
And thus, making
Klaus Darilion wrote:
Is Asterisk not able of handling multiple early dialogs with pedantic=yes?
Asterisk is not capable of handling multiple dialogs in response to an
outbound INVITE at all. The code is not prepared for requests that it
sends to be forked by a proxy.
The next major version of
Fernando Lujan wrote:
Hi guys, I'm trying to use asterisk with my slackware 10.2 box.
Kernel 2.6.13 from the testing...
The udevd are not creating the /dev/zap devices.
Someone already have success installing asterisk over slackware?
Thanks in advance.
Fernando Lujan
I also use asterisk on
Hey all!
I've got my Asterisk box tied into my PBX.
Currently, if a call comes into my PBX, and can't find the extension, it
forwards it through my Asterisk trunk to Asterisk.
This works great!
Is there a special dialplan function (or common
usage pattern) that can do the same thing in
On 5/18/06, Benchev [EMAIL PROTECTED] wrote:
I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from
192.168.0.106 What is necessary to put it to work?
I recently lost my setup (bad drive) and I'm now trying to get my setup
back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an
SPA-3000. I can call the phone extension, I can call from the phone on
the SPA to other extensions and I can call out to the PSTN. What I can't
do is to
You will have problems. Physically the card won't fit into the Soekris
case. There isn't sufficient power to provide ring signalling to FXS
ports, which is why the Digium TDM400 card has a dedicated power
connector.
Of course the Net4801 will only transcode a couple of calls at a
time...which
I'll give this one a try, but don't trust me, test it yourself :-)
Of course Asterisk can do it! All you need to do is set up a rule for
matching ALL extensions in the PBX in it's own separate context and
include that context into your normal context. In the following
example, asterisk will
On 5/17/06, Marcel van der Boom [EMAIL PROTECTED] wrote:
We had the exact same problem. It started happening for us starting at
the 'k' release of bristuff (i mailed a msg on it in february i think
to junghanns).
Marcel, thanks. This does seem to work indeed! I just tested this on
our
Hi
Im new to asterisk and want to setup a small system at home to play
with.
Can anyone advise a good card I can use so the asterisk box Im building
can act as a gateway to PSTN using my single home analogue phone line.
Kind Regards
Kev
___
I know this may sound like a stupid question but I will put on my flame
retardant suit and ask anyway.
Is there any way to use/allow SIP reinvite and still track the length of the
call?
I realize that the whole idea of reinvite is that it takes the proxy out of
the media path which, from
Philipp von Klitzing wrote:
Hi there,
I've been playing with a SNOM 360 and 190 trying to get them talk to each
other using g722 with 16 kHz. However all I see in the SIP log codec
negotiation is g722/8000 which makes me believe that this is only a 8
kHz link (and that's what it sounds
Low end server with a free PCI slot, Asterisk @ Home, Digium TDM11B (1FXO /
1FXS)
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m -
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote:
Otherwise the Diva server cards
are a good option (extensive, but come highly recomended from most that
I hear). Good luck and happy hunting.
Ouch, you weren't joking. 1453 Euro!
What about the Gerdes Primux Cards. They can be used
Mojo Jojo wrote:
Is there any way to use/allow SIP reinvite and still track the length of
the call?
This is discussed nearly every week on this list, it's well covered on
the wiki, and various other places. Have you tried to research this
before asking here?
The simple answer: you are
Andrew Kohlsmith wrote:
On Thursday 18 May 2006 08:07, Darrick Hartman wrote:
Don't attempt to use a Digium or other FXO and FXS card on a soekris
board, especially if you plan on using a full-featured distribution like
CentOS (which is what AAH uses). See more below...
I read below,
Hello,
Does anybody know of free Java/ActiveX applet which could be
placed into web and configured to check ping/latency/jitter/ports to selected
server?
Something like http://www.testyourvoip.com
Regards/Pagarbiai,
Mindaugas Kezys
Give http://www.asteriskguru.com/tutorials
a try.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Adams
Sent: Wednesday, May 17, 2006 5:19
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] New To
Asterisk - Advice needed
Hi
People,
Im
Egads. That'a shame, because from experience I can tell you that trying to make
Asterisk work in a 'hosted' manner is really tricky. Asterisk wasn't designed
with multiple companies in mind.
-Original Message-
From: Damon Estep [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 17, 2006
From the picture on the web site it looks like it uses a cologne chipset.
Any idea if these cards will be available in Australia?
Craig
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks Cosmin!!
I didn't realize that the dialplans ran in sequential order. I'll try that.
thanks!
--Aaron
- Original Message -
From: Cosmin Prund [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May
On 22:32, Thu 18 May 06, Craig Guy wrote:
From the picture on the web site it looks like it uses a cologne chipset.
Any idea if these cards will be available in Australia?
Can't you just order them from the digium website?
Or is digium not shiping to Australia?
--
Michiel van Baak
On Thursday 18 May 2006 16:32, Ralph Liebessohn wrote:
On 5/18/06, Benchev [EMAIL PROTECTED] wrote:
I'm trying to start with Asterisk, but I could not put 2 softphones to
talk. The asterisk server rejects the connections always when I dial.
May 17 07:49:22 NOTICE[1924]: Rejected
Craig Guy wrote:
From the picture on the web site it looks like it uses a cologne chipset.
Any idea if these cards will be available in Australia?
(Please to trim your replies and not reply in the middle of quoted text...)
The cards will be available through all normal Digium distribution
Hi, all. I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension. Is there any
way to do that? I've tried to RTFM, but I'm coming up empty.
Thanks,
-Ken D'Ambrosio
___
--Bandwidth and
On 4/18/06, Wes Baehr [EMAIL PROTECTED] wrote:
Well this is disappointing. Time to find somebody else...
From: NuFone Operations
Sent: Tuesday, April 18, 2006 3:44 PM
Subject: NuFone Update: DIDs
Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier
snip
I received an email from
Aaron Paxson wrote:
Thanks Cosmin!!
I didn't realize that the dialplans ran in sequential order. I'll try
that. thanks!
We originally had dialplans run in random order, but people found it too
confusing.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and
This is not necessarily Asterisk specific but if I have
Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend
a 9 can I do this via the polycom config? I cant find anything
in the docs.
Bill
___
--Bandwidth and
I am trying to call phone that on meridian My asterisk
connected to meridian via E1.but failed
I think I have routing problem between sip and E1
because I am debuging sip call then I see call going
to [EMAIL PROTECTED]
I just want sip client able to call phones that on
meridian and phones that on
I have not found a way to do this via the Polycom
configs. However, what I do is just ensure that the callerid of an inbound
call is set so that the recorded number on the Polycoms is a valid callback
number (i.e., prepend '9' or '91' depending on the inbound
CallerID).
Regards,
- Brad
Any device to legally connect to the PSTN in Australia must be approved by
the regulatory body. A process that usually costs at least $20,000 and only
allows the permit holder to sell the product for conneciton to the pstn. It
is a very high barrier to entry for the Australian market. There
On 18 mei 2006, at 14:05, Jeroen Zwarts wrote:
I tested the outbound and inbound dialling over BRI, and * hangs up
when it
needs to!
I will have to test some more to see if this little patch doesn't
break
anything else, but so far so good.
I've done a bit more testing and in our install
Hi all, is anyone using the Aastra Powertouch 350 analog (adsi) phone with
asterisk? I cannot get the phone to display incoming caller id... I can see
the CID if I hook up a cheap caller id enabled phone... Only the PT350 is
having problems. Looking all over for docs on this phone, but what I've
Aaron,There are probably plenty of ways to do this, off the top of my head, if you add a 'include = go-to-pbx' context within the context where your Asterisk patterns are, and there is no match, Asterisk will then begin to check the 'include' contexts in order. (It does not even look at them
I'm fairly new to Asterisk myself and I also started with AAH.
Unfortunately I had to remove all configuration files generated by
FreePBX (the GUI of AAH) and started over using http://voip-info.org as
my guide. Configuration files generated with FreePBX make use of
advanced functionality
On 05/18/06 18:45 Sebastian Kayser said the following:
So although the Zap interface is used for both types of external calls
(snom - POST, snom - PSTN) the ringing indication to my snoms fails
for calls to the PSTN.
we've got the following:
E1 PRI --- Asterisk ---+--- FXS Gateway --- Analog
Hi All,
I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful.
Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The
On Wed, 17 May 2006, Rodney G. McDuff wrote:
Hi All
Before I go out and buy a DELL PowerEdge 2850 has anyone had
problems (or any other useful experience) getting a TE411P to work with
it. I also have a legacy TE110P. Has anyone had problems with this combo.
I tried using a TE110P and a
Hi every body.I was intalled the last version of unicall, but the problem persist.This issue can be due to a grounding problem?regards.2006/5/16, acriollo
[EMAIL PROTECTED]:I will try ...Regards.
2006/5/16, Moises Silva [EMAIL PROTECTED]:
A recent version of Unicall has a small bug in tone
I am new to this also. But it is hard to figure out what can be done through a single PSTN line. maybe you can buy the SDK card and then setup IVR menu for have separate voice mailboxes. Weidong
On 5/18/06, Barrass Kevin [EMAIL PROTECTED] wrote:
HiIm new to asterisk and want to setup a small
Hi All, as a followup to my previous posting (re: not getting caller id to
display on a powertouch 350), I've found the following...
If I hook the PT350 up to the PSTN line, caller id is diplayed properly...
If I connect it to our CAS Acess Bank II, the 350 will not display incoming
CID... If I
Jonathan Feally wrote:
I believe I had to do the udev permissions file and also cause udevd
to launch at bootup before modprobe'ing zaptel stuff. Check to make
sure that udevd is launching automatically on bootup and that the udev
rules and permissions are in place.
Thanks all. I have this
I'm trying to put asterisk working with a proprietary pbx system.
I'm doing it using a T1 crossover cable. The pbx system uses the R2/MFC
specification. And the don't inform if it uses cas, ccs, ami or hbd3.
My digium card is flashing a red light.
How can I put this working with the R2/MFC
Jon Scottorn wrote:
Hi All,
I have been struggling with the auto dial out in asterisk. I am
trying to get a call to be auto dialed and play back a message once
the line is answered. So far I have been unsuccessful.
Currently what happens is I have my .call file. I mv it into
Colin was right!
I forgot that AMP dialplan makes intensive use of database keys as AMPUSER
and DEVICE: I understand its logic but migration is postponed after a GREAT
dialplan rewriting :-(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
GibbsSent: May 18, 2006 11:37 AMTo: Asterisk Users Mailing
List - Non-Commercial
Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. I'm using a GNet VP168I (same as the PA168V) and it works fine even behind a NAT which is itself behind a corporate
Hello,
I have a client that needs 20-26 simultaneous voip connections and I don't
want to relay all this traffic. So I m looking for a router with non
symmetric NAT for SDSL. (to use STUN)
Thanks
Laurent
___
--Bandwidth and Colocation provided
I'm a bit confused about how to handle this.
I have Asterisk sitting in the middle between a Qwest Long Distance T1
(Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic
D/240SC-T1 card.
The Qwest T1 originally was connected to the Dialogic card directly. The
signaling was
On Thursday 18 May 2006 15:01, Chad Osmond wrote:
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
Yuck. While technically correct, both of those solutions suck major goat nad.
-A.
When used in TDM mode, the sangoma cards will work under zaptel, so you will
need to perform SNMP at a higher level (i.e in Asterisk).
David Yat Sin
Sangoma Technologies
(905) 474 1990 x119
(800) 388 2475 x199
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Wiki: http://sangoma.editme.com
Stupid not-quite-an-answer - if you're willing to pay money for a fix, why
not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will
traverse NATs without a problem.
Looks like it wasn't a NAT of configuration problem after all... the
SPA devices are quite nice, IMO. If there's a
On Thu, May 18, 2006 at 10:38:14 -0600,
Jon Scottorn [EMAIL PROTECTED] wrote:
Hi All,
I have been struggling with the auto dial out in asterisk. I am
trying to get a call to be auto dialed and play back a message once the
line is answered. So far I have been unsuccessful.
Hi all
Which command do I use to pull the mISDN number from an incoming call.
--
Cheers
Wayne
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Has anyone seen good scripts or documentation on Voicemail
groups? We are looking to have a system where you can send a voicemail to
multiple mailboxes.
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