On Mon, 2005-09-26 at 17:24 -0400, Nana Tandoh wrote:
> Hi All,
>
> We are using SER/Asterisk, it works fine from X-lite to corded phones
> but have problems using a cordless phone on the Handytone 496. Has
> anyone experienced this problem.
Well, if you told us what the problems are perhaps we
Hi Marco,
As far as I can recall, the IBM setup utility can enable you to change the
IRQ of the SCSI controller.
In addition, I've never seen a WildCard board bound to IRQ7 on any box,
which is very weird in it self.
I'm flying over to Ireland today (actually, at the airport right now), and
I'm
In order to use a with GrandStream BT-488 as "pass-through" gateway, I
need a way of sending the FXO port off hook when I'm using the FXS port
for VoIP communications, because I want to use the "hunting line" feature
to let incoming call skip that FXO port and move on to the next free line.
The onl
I have an * box connected to a Nortel SL100 through a PRI (US) using the
Digium TE410P (quad-span T1 card). I don't have access to the SL100 -
it is handled by another group.
The span comes up OK (timing, framing fine). However, as soon as the
D channel comes up, I get endless "HDLC Bad FCS" err
If you are willing to dedicate a fax line and forward faxes to a
dedicated extension it work 100% for Incoming and Outgoing faxes with
Sipura-3000
Though, you have to change in the Regional Tab:
Ring Waveform: from Sinusoid to Trapezoid
I've been using NVbackgroundDetect with Sipura-3000 and for
I'm afraid this may not be helpful, but I will try,
When I was working at a previous company, they where just starting to switch
everything over to VoIP, since I was the new guy, I got one of the VoIP
lines because of not having any more "real" phone lines.
I had the same problem, the Asterisk gu
You may interested to know that a lot of connections will need inband when
speaking server to server. My system runs all users on RFC2833, except for
other asterisk servers. They run inband, because otherwise the DTMF wasn't
working.
Just my 0.02
->-Original Message-
->From: [EMAIL PROTE
On Mon, Sep 26, 2005 at 10:00:10PM -0400, Patrick wrote:
>
> I know it seems basic but did you make sure and plug power into the
> board when you installed it into the PCI slot? I spent about three hours
> trying to get the dang thing to work in my machine until I decided to
> stick the card
Can anyone recommend
a good IAX provider offering numbers in Toronto and
Detroit?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asteri
Does anyone have any experience with Teliax for inbound IAX?
Jason
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBS
Tim
We have used a SPA3000 with asterisk 1.0.9 and an E1, recieves fine, sending
is impossible.
Regards
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Litwiller
Sent: Tuesday, 27 September 2005 2:08 PM
To: asterisk-users@lists.digium.com
I've been running with a generic X100P for 5 or so months and every once
in a while I have problem receiving faxes. I see that others have the
same problems and some worse than I have with these boards so I was
wondering if using a Sipura SPA-3000 would be any more reliable.
Has anyone had en
>From Show application stripMSD
StripMSD(count): Strips the leading 'count' digits from the channel's
associated extension. For example, the number 5551212 when stripped with
a
count of 3 would be changed to 1212. This app always returns 0, and the
PBX
will continue processing at the next priori
Dear list,
I have trying to work with the ICD with Asterisk
Strange i could not able to even compile it .
I have followed the ICD readme but no use ... ?
And the ICD list in sourceforge seems to not much active.. ?
Any suggestions are welcome
with regards
rk
Yahoo! India Matrimony
Dear list,
I want to use the asterisk fifo channel to be integrated with other applications in the asterisk like MoH .
Do we have any implementations as such
Basically what does we mainly use this asterisk fifo ?
your views will be highly appreciated.
with regards
rk
Yahoo! India
For years we've had the following simple context for outgoing calls:
[outtrunk]
; match any NANP, and strip leading 1 off
exten => _1XX,1,StripMSD,1
; dial outbound on trunk group 1
exten => _XX,2,Dial,Zap/g1/${EXTEN}
But when I upgraded on Friday to the latest CVSHEAD, this no l
Ladies and Gentleman,
Could someone point me in the direction AMP add-ons? I havve the
ones that were referanced in the AMP instal PDF, but I cannot find the
| Maintenance | addon like is found in [EMAIL PROTECTED]
Does the Panal in AMP tell you anthing or do you have to some how login
to it bef
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 24, 2005 2:30 PM
To: Gregory Wiktor - ADCom Corp.
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one
msn
On Fri, 23 Sep 2005 [EMAIL PROTE
It would be prudent the test for success and continue rather than
failure and drop.
For example:
exten => s,5,GotoIf($["${SYSTEMSTATUS}" != "SUCCESS"]?105:6)
That way only the result that you know is good, Will continue a call..
> -Original Message-
> From: [EMAIL PROTECTED]
> [ma
Hi Yair,
Please let me if you managed to fix the DTMF tone issue, which you were
experiencing couple of months ago. If not can you share any advancement.
I'm currently experiencing the same issue, I can make outbound calls but
DTMF will not work when dialing IVRs. My configuration is [EMAIL P
On 2005-09-26 at 18:15, Jim Gottlieb ([EMAIL PROTECTED]) wrote:
> But since (as far as I know, without using AEL) there is no conditional
> branching based on a variable, how am I supposed to use this?
OK, I forgot about GotoIf. However, the doc is wrong (or at least
incomplete), because it only
You may loose 'control' of the call but you can always 'get it back'
Use the UnigueID of the call to track it throught Asterisk. You can
palce a monitor event to redirect, bridge, drop, answer or antything
else.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED
Try the following:
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Read(PIN,87)
exten => s,4,System(/usr/local/bin/pin -c ${PIN}) ; check it
exten => s,5,System(/usr/local/bin/pin -d ${PIN}) ; delete it
Exten => s,6,GotoIf($[${SYSTEMSTATUS} = FAILURE]?105:7)
exten => s,7,SetAccount(${PIN})
I know it seems basic but did you make sure and plug power into the
board when you installed it into the PCI slot? I spent about three hours
trying to get the dang thing to work in my machine until I decided to
stick the card into another PCI slot. That is when I noticed that I had
forgotte
We upgraded to the latest version of asterisk (because we needed some
newer features), only to find all our PIN applications accepting any
number the caller makes up!
I traced this to the System application completely changing the way it
deals with success or failure of the program it calls.
Prev
We are having
problems with the above configuration. A Canon fax in a SPA3000, Asterisk
1.0.9 stable with single Digium E1 card. Fax's are received well but
sending is next to impossible. Anyone seen or heard of any solutions to
this type of issue?
Regards
Mark
ACC Australia
Phone:
I've used Biostar U8668D and Asus P4GE-MX.
Chris
- Original Message -
From: "Brent August Torrenga" <[EMAIL PROTECTED]>
To:
Sent: Monday, September 26, 2005 6:09 PM
Subject: [Asterisk-Users] Socket 478 Motherboard for use with TDM400P
> Well, it has been a long saga over here try
I'm getting unstable behavior with my newly installed TE110P T1 card. It
hangs up any incoming call anywhere from 20 seconds to 6 minutes.
Frequently, when you call back on our incoming T1 there'll be an
automated announcement (maybe from the telco?) stating "we are unable to
complete your cal
Well, it has been a long saga over here trying to get a TDM400P to work
with an Intel D845HV motherboard, and the towel is close to being thrown
in. Does anyone know of a (few) motherboards that work well with the
TDM400P and are socket 478?
--Brent
___
Greetings,
1.) Voipbuster does not support T.38. If you can get a clean connect
using G.711u then the answer is maybe. Latency will ice a analog
connection.
2.) That's built into the dialplan at VoipBuster. It's doubtful they'll
remove the "routing charge" message, but you could always ask the
Hi,
I'm using voipbuster at work, and I've got 2 questions:
1) Is it possible to send faxes using voipbuster connex?
2) Is it possible to cut off or cover the voice that say the charge
per minute?(I've payed the '5' euro, and from that moment I've got
it!).
Of course I understand that is to let me
Of possible interest to people having various issues with TDM400 cards,
is that a fix has just been submitted to CVS for the issue where CPU
usage would regularly spike up to 100% with the wctdm driver loaded.
Regards,
Richard
___
--Bandwidth and Col
On Mon, September 26, 2005 20:35, Asterisk said:
> hi Asterisk users,
>
> I am in the UK and trying to get an asterisk system running.
>
> I have the SIP side of things running or limping along to the best of my
> newbie
> ability.
>
> I have a problem with a FXS card. Connecting a standard (Worki
Hi guys,
does anybody succesfully connect Asterisk with netappel?
I've tryed using voipbuster settings, but doesn't work.
Any suggestions, are wellcome.
Thanks
--
.:FaberK:.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mai
Hi!
About realtime...
Anybody knows a unixodbc driver for oracle (free or comertial)?
I am working with a trial easysoft odbc-driver, but the commertial license
is very expensive...
Another question.
res_odbc.so works with IODBC ?
Regards.
Jsalas
___
Trying again...
*Summary:*
I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE
mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode;
What card(s) should I put in to these servers?
*The long story:*
I have 3 locations I want to connect using (*) servers.
1 of
On Mon, Sep 26, 2005 at 07:17:36AM -0400, Andrew Kohlsmith wrote:
> On Monday 26 September 2005 00:36, Kevin P. Fleming wrote:
> > Bridged calls with 2nd gen firmware result in the audio never leaving
> > the card; that's why you are seeing such an improvement. Essentially,
> > the Zaptel 'native b
Hi, I am using AsteriskJava and I have some problems, I will appreciate
any help...
My system has the following architecture (in the server side):
- An app server (connected to the asterisk console)
- An AGI Server (developed with AsteriskJava)
- An AGI Script (executed by the above AGI Server)
OK I have just gone live with asterisk in a new office with approx 40
Polycom 501 handsets. I have a few questions:
1) Call Parking: I am able to park calls using the standard Asterisk
call parking system (transfer to ext *70 etc...) I would like to make
this a little easier for my users Th
I am running Asterisk 1.0.9 and the latest bristuff in combination with a
HFC isdn card, connected to a BRI interface.
For some reason, I am not able to have it dial out (see below). It exits
with DIALSTATUS=CHANUNAVAIL.
One thing that may be misconfigured is that it says: Signalling Type: PRI
Sign
Using voicepulse retail account because I need the caller ID name. Maybe
they have added that to connect accounts by now.
Jason Schafer wrote:
This sounds like a winner, are you using voicepulse?
Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs
that are fixed? I don't
Hi All,
We are using SER/Asterisk, it works fine from X-lite to corded phones but have problems using a cordless phone on the Handytone 496. Has anyone experienced this problem
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Price is about the only good thing...
quality? Jajajajaj
reliable? Jajajajja
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Schafer
Sent: Monday, September 26, 2005 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Hello,
I have two extensions 100 and 200 and I need in "h" context dial first
extension by asterisk and when the extensions
100 is answered, then to dial exten 200 and then bridge them together..
How can I do that?
I need it to "call back" when called party is busy.
thanks
__
Hi, Why the command "sip show objects" isn't in the version 1.0.9? Exits any
similar? Thanks. Ezequiel
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mai
I have got Areskicc installed with AMP and I can’t
stress out how good and excellent this software is.
I must say the author desire every credit for this software and I would like to
say Thank you to Arezqui who wrote the software.
Now there is one last thing before my whole asterisk c
Hi all,
Just a couple of quick questions. I have a HP DL360 G4 (dual Xeon EM64T 3.0Ghz
processors). I am using a TE411P in the
system.
1. Should I run the a x86_64 Linux (CentOS) or just go with the plain
old x86 version? Is there any benefit (or things to be aware of) on
x86_64 vs x86?
2.
This sounds like a winner, are you using voicepulse?
Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs
that are fixed? I don't really like to upgrade unless I need to.
Jason
Paul wrote:
connect.voicepulse.com allows up to 4 calls at a time coming into an
$11/month DID w
Nathan Pralle wrote:
I am trying to enable dial-by-email by using LDAPget to query an
Active Directory server. I've got it retrieving the phone number
fine. Unforunately, the numbers stored in active directory are
either in the format: (xxx) xxx- or xxx-xxx-. Is there
Chris
I have only ever used zaphfc drivers and for me they are perfect. Echo
has never been a problem. It would be helpful if you were to provide a
bit more information to the group about your configuration so we can try
and help you work out the cause.
Switching to capi or mISDN is unlike
trixter http://www.0xdecafbad.com wrote:
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote:
This can be done by modifying the source code.
how helpful. If I modify it enough it will be 100% identical to windows
xp, anything can be done by modifying any code. That however d
Anyone know if the INTEL/Dialogic announcement will become available to
us who do not use the asterisk BE?
Just curious as I have used those cards in the past and they are VERY
STABLE and VERY dependable with excellent quality.
~ron
___
--Bandwidth a
Brian C. Fertig wrote:
yes.. I have looked. they are different. But when I unregister 1 the other will register..
Its only when I have 2 of them trying to register at the same time I have an
issue. But yes
the ID's are different in both of them.
maybe you have the same aliases
_
I am trying to enable dial-by-email by using LDAPget to query an
Active Directory server. I've got it retrieving the phone number
fine. Unforunately, the numbers stored in active directory are
either in the format: (xxx) xxx- or xxx-xxx-. Is there
any way to parse
connect.voicepulse.com allows up to 4 calls at a time coming into an
$11/month DID with choice of IAX or SIP
there was discussion here and other places before about broadvoice
allowing the calls but then charging 3.9c a minute for the extra
channels used
shop carefully
Jason Schafer wrote:
Hi,
I asked yesterday about a problem with x306 and IRQ sharing, didnt get
much info, now, i was playing with lspci, and see something strange,
lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is
also on IRQ 7,
lspci -bv (from the man - b - shows "bus-centric view, as seen by
IMHO - you should not use price and quality in the same sentence for BV.
On Mon, 2005-09-26 at 15:20 -0400, Jason Schafer wrote:
> I'm relatively new to the whole VOIP game, here's what I want to do. I
> am using VOIPJet for all of the outbound calls on our AAH box. I have
> one landline that
Not a bad idea, thank you for that. I'll look into
it
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ
WeschkeSent: Monday, September 26, 2005 2:37 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Call Back On Bus
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote:
> This can be done by modifying the source code.
>
how helpful. If I modify it enough it will be 100% identical to windows
xp, anything can be done by modifying any code. That however doesnt
answer my question with anything that isnt
hi Asterisk users,
I am in the UK and trying to get an asterisk system running.
I have the SIP side of things running or limping along to the best of my newbie
ability.
I have a problem with a FXS card. Connecting a standard (Working) UK phone makes
the phone ring all the time while on hook.
Andrew Kohlsmith wrote:
This is why so many of us are pushing Digium to PLEASE FOR THE LOVE OF GOD
print a detailled list of what's improved with the new firmware... None of
us have any clear idea of what has changed from v1 to v2 and little things
like this are unbelievably important.
The
I'm relatively new to the whole VOIP game, here's what I want to do. I
am using VOIPJet for all of the outbound calls on our AAH box. I have
one landline that I would like to busy forward to an inbound VOIP
number. Broadvoice was recommended to me for price and quality.
Can anyone make a su
I tried for weeks with an AB I, and never got anywhere...I could not get the T1
to sync properly. I switched exclusively to ADIT 600's and have had no issues
since.
-Darren
From: [EMAIL PROTECTED] on behalf of Time Bandit
Sent: Mon 9/26/2005 2:25 PM
To: ast
That solutions kind of works, but only for
phone numbers that are in the format xxx-xxx-. The records I am receiving
from LDAP vary significantly because the user can control their entry. So I
can receiving mostly the following formats…
(xxx) xxx-
xxx xxx-
xxx-xxx-
Uppal,
Can you guide me through in seting up trunk, ratecard etc.
Goksie
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Junaid Uppal
Sent: Sunday, September 25, 2005
12:19 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [
Well, you need to be a bit more specific. How are you trying to send
it? Are you using an SMSC? What kind of lines do you have?
--
Tom
On 9/26/05, Jerry Geis <[EMAIL PROTECTED]> wrote:
> Does anyone know about sending SMS messages to a sprint pcs phone.
>
> Can you give me a few details. Thanks,
I had similar problem using Digium TE406 card.
Try update the driver. I worked for me.
Good luck.
AK
On 9/23/05, maka <[EMAIL PROTECTED]> wrote:
Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode.
The ISDN ca
more information about the hint priority in the extension file:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extensions
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20BristuffDevstate
> I have a client that has an old Merlin system. They would like to move to
I have had good luck with a php-based status webpage. My users don't
mind having a page up in their browser. If you were to use Soundpoint
IP600s, this dynamic page could be shown on the phone, which IMO would
be great unless you had lots of users to monitor.
Also, the polycom soundpoint lin
Capt MS wrote:
thanks for the reply
Is Digium card compatible with EPABX standards
available in india , further how much does a card with
three FXS and one FXO interface cost,
Do u have any experience of implenting the same ,
I am in army what we lookin at is voice gateway to
interface our PBX
I have to replace a custom PBX, that is infront on a IVR system based on OLD
NMS AG-E1 Card.
The Cards is configurated with CAS Digitalmode, someone can give me some info
about Digim Cards CAS configuration i need a conversion Table?
I wanto to don't touch configuration on winbox, i want only
Is there a functional reason why you'd use MeetMe here? I think probably the easiest way to accomplish this is to use an DeadAGI script which can be invoked via the 'h' extension in the context that would then perform the functionality you're looking for and if they get through it should just brid
On Snom phones this feature works (look at the "Hint" Command in
extension.conf.)
Support for this should come for the Grandstream GXP2000, currently it does not
working.
Cisco 79x0, i dont know.
> I have a client that has an old Merlin system. They would like to move to an
> Asterisk based sy
FOP does this quite nicely
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
LaroffSent: Monday, September 26, 2005 1:57 PMTo:
Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Extension
availabilty
I have a client that has an old Merlin system.
Darren Wright wrote:
I am also a long time client, and have no incoming BV today.
-Darren
it works here today but they can be a bit unpredictable
I use a cheap byod lite account mostly as a test tool. I figure if they
grow up someday I might use them more.
I have been wondering if they
The snom360 phones along with the current CVS-HEAD of Asterisk can presently do this. You'll want to do a wiki search on "Hint" in the dialplan for implementation details.
Polycom has also just released a DSS sidecar to go with their 601 model phones, but the firmware to support more than 8 app
I posted this 1.2.0-beta1 success story to asterisk-dev, and someone
recommended that asterisk-users might benefit from it as well.
Thanks,
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
-- Forwarded message --
Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT)
Subject: [Asterisk-
> Re: sipuras 841 bad sound (Juan Jose Comellas)
> On Tuesday 20 September 2005 20:46, Anton Krall wrote:
> > I have a problems with some sipuras 841 and asterisk 1.0.9.
>
> (upgrade the firmware was suggested and completed, and didn't fix the
> problem.)
There are a few little configuration deta
Hi,
Is there somebody using an Access Bank I with Asterisk that could
share the secret ingredients needed to make it work ?
I've searched around and found some info, I tryed almost every
configuration possible but I can't seem to find the right combination.
If someone could provide me with the co
Have you tried:
[EMAIL PROTECTED]:[EMAIL PROTECTED]
[EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]
[EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/05075034132
?
Sometimes SIP providers require the realm in the username, so the first part
should have the @blah
Then, the third part, is the callerid so it
Does anyone know about sending SMS messages to a sprint pcs phone.
Can you give me a few details. Thanks,
Jerry
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digiu
Thanks for the information Sherwood.
Then the question I had if the normal routing works for the SIP proxy
works with a LAN server.
But I cant get a success in connecting the router LINE1 to Asterisk.
WRT54GP2 says as status "Can't connect to login server" and there is no
connection attempt when r
Anyone else out there have some thoughts? The customer
wants to be able to control what can be redialed on busy, such as no
international. I'm having my doubts as to whether or not this can be done. My
idea seems like it would work, but after the customer hangs up, wouldn't the
context stop
I haven't received any responses. Just wanted to follow up and see if anyone has ideas?
It seems like there ought to be a way to do this, especially since the
TDM400 FXS card is able to send the proper signal to the connected
phone. It seems like there just needs to be a way to configure the FXO
c
I have a client that has an old Merlin system. They would like to move
to an Asterisk based system, however, with their existing system
each phone is capable of displaying who is on the phone within there
office. This is done by lighting a red light for each line(extension)
that is in use. Has any
Are you using Digium's new v2 firmware? If not I would recommend
against it. I currently have 2 Sangoma quad T1 cards in a single server
and it works just fine.
Previously I had 2 TE405P(with old firmware) in the machine
and had interrupt issues. Replaced with Sangoma boards before Digium v2
fi
Are you using CCM to operate your gateway with MGCP? If so, I had to
change the default timers under CCM advanced setup for "Media exchange
timers" or the call was timing out at 4 seconds. If the setup was
complete prior, it worked fine, but after 4 seconds q.931 from CCM would
tear down the call
Waldo,
Thanks for the information. If you don't mind answering: are you guys
developing this solution for your internal needs (meaning serving UAs
from within your enterprise) or are you planning on offering services
to the public?
This solution is being developed for our internal needs.
It
Kevin Scott wrote:
I'm not sure what he/she was sending as the caller ID information, what I
was trying to do, was send a normal 10 digit number as caller ID. Is there
any solution to this? Or anything planned?
There are no plans to allow just any caller ID to be sent. Once US dids
are ava
Hello,
I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:
As you can see belo
I have Asterisk server(1.0.9) behind Iptables firewall.
I configured Iptables and sip.conf as below.
Andrea(2000) is the outsider phone, on Internet with public IP
Luca(2001) is the insider phone, on local network with private IP as well
Asterisk server.
I noted the ports in play are 5060, 8000, 8
I have been trying on and off for a couple of weeks to no avail...
Darren Wright wrote:
I am also a long time client, and have no incoming BV today.
-Darren
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation
Has anyone noticed that a ? Entered at the root CLI does not work any
longer?
Petty I know but I did use it.
--john
--
This mail was scanned by AntiVir Milter.
This product is licensed for non-commercial use.
See www.antivir.de for details.
___
--Band
I'm not sure what he/she was sending as the caller ID information, what I
was trying to do, was send a normal 10 digit number as caller ID. Is there
any solution to this? Or anything planned?
Thanks for your time,
Kevin
___
--Bandwidth and Colocation
Hi Ade,
An example of oh323.conf is attached, and the lines in the extensions.conf
that make the choice os this oh.323 channel is:
[globals]
GK => OH323/IP of your GK
[local]
ignorepat => 0
exten => _0021NXXX,1,Dial(${GK}/${EXTEN:1})
exten => _0021NXXX,2,Congestion
--
[ ]'s
Da
This can be done by modifying the source code.
trixter http://www.0xdecafbad.com wrote:
I have been asked if asterisk can change codecs dynamically based on the
calling party's caller id. I couldnt find anything, and dont know that
this is something that asterisk can do, but it occurs to
Yes indeed. There have been huge changes to chan_zap.c in CVS-HEAD compared
to 1.09.
In 1.09 Stable there are a lot of problems with handling call hang-ups.
CVS-HEAD, of 28/08 was much better. But even though it did improve things,
it wasn't quite right. In particular I found two problems with pol
I am also a long time client, and have no incoming BV today.
-Darren
From: [EMAIL PROTECTED] on behalf of Jason Schafer
Sent: Mon 9/26/2005 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and Broadvoi
El jue, 22-09-2005 a las 19:04, David McNett escribió:
> I made http://slacker.com/~nugget/stuff/asterisk-cow-real.bmp for my
> non-Linux asterisk servers.
I made my * + tux + office logo
http://www.cipher.com.pe/central/asterisk-tux-cipher.bmp
Regards,
--
Ing CIP Alejandro Celi Mariátegui
<[
Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk%
20Cisco%20CallManager%20Integration ?
I've followed these steps and I can make calls from a CCM client to
Asterisk, but the end point at the Asterisk side can't hear any audio.
On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman
1 - 100 of 166 matches
Mail list logo
