Hi,
this blinking problem of yours occured to me and it vanished when I told
Hicom to be the master of the line and * the slave.
How exactly do I tell asterisk to be the slave?
AFAIK signalling=pri_cpe in zapata.conf means, that asterisk is the slave,
and signalling=pri_net means master, or
hi friends !
can anybody tell me something about cdr configuration.
actually i want to confirm about the minimum requiremnts.
is it possible to configure it with mysql server and myodbc anly or unixodbc
is also required?
in case unixodbc is also requied than help me to send some download links
Hello Deepak,
yes, you can use mysql. the packages are in asterisk-addons.
there is a very good wiki page on the subject here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
hope this helps,
yair
On Apr 6, 2005 7:33 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
hi friends !
can
I just had a quick question. I was wondering if it is possible to have *
register itself with a Cisco CallManager using SCCP. In other words it is
possible to have a SCCP trunk?
Thanks,
Jared
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Hi,
-Original Message-
I just had a quick question. I was wondering if it is
possible to have * register itself with a Cisco CallManager
using SCCP. In other words it is possible to have a SCCP trunk?
Nope. Hasn't been implemented yet AFAIK.
Florian
Hi
Newbie asterisk guy here and forgive
this slightly long mail, but I'm stuck on
this for a week.
I'm having major problems getting a
Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing
out). I'm getting the 0x3301 or 0x3302 error.
My capi.conf
Hi,
what configuration on asterisk is missing when SJphone tells me that the
voicemail number is not configured?
current config:
sip.conf
[EMAIL PROTECTED]
voicemail.conf
102 = 102,Kib, [EMAIL PROTECTED]
What is missing?
Regards,
Kib
___
I installed Asterisk in a default way. I ran over many manuals and
FAQ's on asterisk.org. However, I found that many exaples included in
them were equipment-dependent. I do not know how to configure my
Asterisk for my X-Lite.
Is anybody willing to help me?
Regards,
Abe
Roy Sigurd Karlsbakk wrote:
1.Does Asterisk support SS7 and ISDN?
ISDN is supported out of the box. SS7 support is (or will soon be?)
supported by a commercial version of Asterisk. Search the list
archives or
post to asterisk-biz.
Steve Underwood (here on the list) has a commercial ss7
Hello!
Does anybody tried to work with IAX2 (client side - softphones) behind a
NATs that always increment ports?
At asterisk CLI I see:
-- Registered '12345' (AUTHENTICATED) at a.b.c.d:22269
-- Registered '12345' (AUTHENTICATED) at a.b.c.d:22289
-- Registered '12345' (AUTHENTICATED)
Why dont you just get the netweb phone which already has iax
support
On 5 Apr 2005 at 17:13, Sean Kennedy wrote:
denon wrote:
Hi all,
I've put together a quick petition, in hopes that we can possibly
persuade Sipura (or any other large-scale IP handset manufacturer) to
include
I beleive the Asterisk example config has an example, but this works
for me (Xlite-linux-beta):
[martijn]
secret=secret
type=friend
context=from-sip; Where to start in the dialplan when this
phone calls
;callerid=John Doe 1234 ; Full caller ID, to
Looks like what cisco uses for their Call manager and PIX stuff .. hahah
.. sweet thanks for the link.
On Wed, 6 Apr 2005, Paul Hales wrote:
Another option - these 1U short depth boxes...
http://www.spinserver.com/
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Anyone tried this on Virtuozzo?
/Danny
On Mon, 2005-04-04 at 22:04 -0700, snacktime wrote:
I just got a linode account and got * up and running without any
problems. I was going to ask them to load zaptel/ztdummy, but I was
wondering if anyone else was interested in an * friendly UML hosting
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stefan Tichy
Sent: Tuesday, April 05, 2005 11:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: mISDN + chan_misdn and DTMF
On Mon, Apr
I modified /etc/asterisk/sip.conf to add my X-Lite soft UA, which is
assigned the user name 177210. Now I want to add another UA with a
user name 177209. Well, asterisk is running. How can I make the
changes to sip.conf take effect ?
Any good idea?
Regards,
Abe
sip reload
or just reload
On Wed, 2005-04-06 at 16:31 +0800, Abraham WEI wrote:
I modified /etc/asterisk/sip.conf to add my X-Lite soft UA, which is
assigned the user name 177210. Now I want to add another UA with a
user name 177209. Well, asterisk is running. How can I make the
changes to
Hello,
I'm trying to get feed back from other Asterisk users of
Welltech WellGate 3701A / 3702A
Or Micronet SP5012s / SP5014s
Or Immix Tel C3-FXS/FXO
Or Euro Teletech VIP-400
(All those are in fact the same product...)
Trying to find/share ideas/comments about registrations issue, caller ID
Laurent Foulonneau wrote:
Hello list,
Does anyone know about a web/php interface to deal with users in
Realtime's Mysql database (sipusers and sippeers tables) ?
Thanks in advance
Laurent
PhpMyAdmin
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phpmyadmin :)
Matteo.
Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha
scritto:
Hello list,
Does anyone know about a web/php interface to deal with users in Realtime's
Mysql database (sipusers and sippeers tables) ?
Thanks in advance
Laurent
Hi
Repeated e-mail as I forgot to make plain text - sorry. Newbie asterisk guy
here and forgive this slightly long mail, but I'm stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing out). I'm
On Tue, 2005-04-05 at 21:44, Carlos Rojas wrote:
Hi
here there are some of them
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files%20international
Hehehehave you tried these out from /usr/src/asterisk ?
%tt-monkeysintro.gsm%They have been carried away by monkeys
Hi
Repeated e-mail as I forgot to make plain text - sorry and then forgot subject
line. Newbie asterisk guy here and forgive this slightly long mail, but I'm
stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been
Altus Snyman wrote:
Good day all
I have a Fedora core 3 installation
Is there any hassles with asterisk?
Thanks
Altus
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On Wed, 2005-04-06 at 10:58 +0100, Paul Redstone wrote:
I'm having major problems getting a Fritz card to dial out in the UK (or
indeed
answer, but I've been concentrating on dialing out). I'm getting the 0x3301
or
0x3302 error.
--
The 0x3301 and 3302 errors seem to
Hi all!
Who can tell me what the correct/preferred/only DTMFmode setting is for
Windows Messenger SIP clients?
Regards,
Evert
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[EMAIL PROTECTED] is believed to have said:
I installed Asterisk in a default way. I ran over many manuals and
FAQ's on asterisk.org. However, I found that many exaples included in
them were equipment-dependent. I do not know how to configure my
Asterisk for my X-Lite.
Is anybody willing to
Hi,
my brand new wcte11xp works like a charme of first boot, then if I
shutdown -r now
the server is not detected at reboot (no such device after modprobe).
Turning off the pc and cold restarting fixes the problem.
Has someone experienced such behaviour before ?
Tnx for any help!
--
I am adding an extra AVM Fritz card to an existing setup..
I have followed instructions from
http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
I did a make clean before I started, made all the changes as specified on
the
Hello,
I connected the TE110P directly to the carrier port now.
I got a green signal light at the card, zttool reports no problem.
Still, I can't dial out.
Asterisk reports:
chan_zap.c:7143 zt_pri_error: PRI: !! Got S-frame while link down
When I try to dial out with a SIP phone:
hi friends !
my asterisk is giving one error while running.
it says that unable to write audio data, module codec_speex.so is not
loaded.
have anybody face this kind of problem than plz tell me the solution.
thanks
Deepak Dhiman
___
Asterisk-Users
We target Sipura because they are relatively a small company, the core
developers at Sipura used to work for Cisco and worked on their ATA product
before they started their own company. A small company is much more likely
to try something new with little lead-time. Also access to decision-makers
If your voicemail is setup correctly then you need to klick on the options icon
on your sjphone and go to profiles. Choose the profile that you are using and
click edit. Then go to General and type the extension that you use in the
voicemail address field.
Cheers
Rikard
-Original
IPSwitchBoard
Version 0.75 has just been released, it has three additional languages:
* Spanish
* Dutch
* Romanian
Download from
here:
http://ipswitchboard.thorben.dk
IPSwitchBoard is
an Operators
I'm having major problems getting a Fritz card to dial out in the UK (or
indeed
answer, but I've been concentrating on dialing out). I'm getting the 0x3301 or
0x3302 error.
Not a direct answer, but are you _absolutely sure_ the card works? I
had the exact same thing late last year here in
Thanks, I entered the voicemail number to SJphone clicked the mailbox
button. The programm dials but nothing happens.
Rikard Westlund wrote:
If your voicemail is setup correctly then you need to klick on the options icon
on your sjphone and go to profiles. Choose the profile that you are using
Hello Folks,
I am trying to compile zaptel
in a RedHat9 plesk/ virtual dedicated server
environment. I have
root access to the machine, but when I try compile zaptel it complains of the following:
Make Results:
``
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE
Hello all
Anybody had any luck with Snom 190 + NAT ?
I have been searching for the correct settings to use , No luck yet.
Also I have been told that I might need to implement an SBC (Session
Border Controller). Yes I have got a STUN server running , but the snom
does not seem to use the random
CuPoTKa wrote:
Hello!
Does anybody tried to work with IAX2 (client side - softphones) behind a
NATs that always increment ports?
At asterisk CLI I see:
-- Registered '12345' (AUTHENTICATED) at a.b.c.d:22269
-- Registered '12345' (AUTHENTICATED) at a.b.c.d:22289
-- Registered '12345'
Min Hwan Chang wrote:
Evening,
I'm having problems with a Polycom IP300 giving me a Stopping
Retransmission Found:102. It gives this error about every 30
seconds.
After searching the Help list, I went ahead and set Disallow=all and
allow=ulaw. This still doesn't seem to help.
Is this problem
Eric Wieling aka ManxPower wrote:
This is the way NAT works. It's not a problem for Asterisk unless you
are doing something silly like port forwarding 4569/UDP on your NAT
router. Asterisk doesn't CARE about the source port of the client.
Yes... but, doesn't it loose connection? For example,
Hello all
Anybody had any luck with Snom 190 + NAT ?
I have been searching for the correct settings to use , No luck yet.
Also I have been told that I might need to implement an SBC (Session
Border Controller). Yes I have got a STUN server running , but the snom
does not seem to use the
I'm having major problems
getting a Fritz card to dial out in the UK
(or indeed answer, but I've been
concentrating on dialing out). I'm getting
the 0x3301 or 0x3302 error.Not
a direct answer, but are you _absolutely
sure_ the card works? Ihad the exact
same thing late last year here in
Hi,
I`ve installed on FC-3 last month and its working gr8... no probs so far
Imran
On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I have a Fedora core 3 installation
Is there any hassles with asterisk?
Thanks
Altus
Thanks for the trouble
n Wed, 2005-04-06 at 15:00, iMRAN wrote:
Hi,
I`ve installed on FC-3 last month and its working gr8... no probs so far
Imran
On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I have a Fedora core 3 installation
Is there any
Sorry I forgot to say we are running the latest stable version of
firmware.
http://www.snom.com/download/share/snom190-3.56u-SIP-j.bin
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gustavo
Russo
Sent: 06 April 2005 13:53
To: Asterisk Users
Hi,I'm having
trouble running ./configure when trying to build spandsp-0.0.2.Obviously,
I'm missing something simple, however, I just can't figure out what it
is.Maybe I'm missing some package I'm not aware of?Here is the error
from ./configure checking how to run the C++ preprocessor...
Is it possible to connect a display that shows the
costs of a call in progress?
Can I also send the call cost to a grandstream
display?
Greeting
Johannes
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Hi,
Is there some sort of Syntax checker for the Asterisk config files. Also
is the /var/log/messages the only file that Asterisk dumps messages too
or is there another place. My system works but I want to make sure I am
not getting any warnings that might affect stability.
Thanks
Hi,
What ports do I need open on the asterisk server (using an iptables
firewall) to allow my sip phones to still work correctly?
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To
Johannes,
I would be curious to know if there is a solution for this. Another solution is
that you buy a call meter. Which is a small box that can be placed in front
of phone phone and that can display costs.
FXS-- call meter -- analog phone
This call meter needs to be programmed with a table
I'll elaborate slightly more... the wiki says:
# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
# IAX2- the IAX protocol
iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
# IAX - most have switched to IAX v2, or
All,
I added to my dialplan something like
exten = 777,1,SMS(0,s,530,Hello)
thinking this would send the Hello message to my SIP/530 phone
(Hitachi WIP 5000).
The phone is working just fine just not receiving SMS yet. I enabled SMS
on the phone.
I was just wanting to play with the SMS and just
On 2005-04-06, Antoine Delaporte [EMAIL PROTECTED] wrote:
Also, any dangers/performance issues from an isp point of view for
running ztdummy?
Few month ago I've run my Asterisk, under an UML.
I was very disapointed, my Sipura3k don't stay registred much more than
12 hours
I'm not really
Title: RE: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
You can stop trying. They still have problem with the firmware concerning the FXO port. If you really want to make a call from * out the PSTN, I suggest you to get a x100p. They are selling it on ebay for $6.99, and I
I'm having this problem too, with a Swissvoice IP10... No nat between
asterisk and the phone... I don't have any problems with the phone,
outgoing and incoming calls work as expected...
Could it be related to qualify=yes?
Julian J. M.
On Apr 6, 2005 1:39 PM, Eric Wieling aka ManxPower [EMAIL
If it helps, then great !
This is what we do in Africa for some callshops that do not want to pay $$$ in
billing software licenses.
While talking, I wonder whether the field of CAller ID which is displayed in
the IP Phone can be updated while conversation is ongoing, say every 10
seconds. In
Hello All, I have googled this problem and called Broadvoice and I still
haven't found an answer.
I have 4 BroadVoice accounts on one Asterisk box. If you call lines 1-3
it rings on line 4. It makes it hard for routing inbound calls.. If I
change the order of the registered accounts in the
[EMAIL PROTECTED] wrote:
Johannes,
I would be curious to know if there is a solution for this. Another
solution is that you buy a call meter. Which is a small box that
can be placed in front of phone phone and that can display costs.
FXS-- call meter -- analog phone
This call meter needs
-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]
I think there are TE410P compatibility issues with other
motherboards as
well. Google the archives (site:digium.com) under HP
Proliant G4 for
example, as I remember some problems there.
This response from
Richard Dutton wrote:
I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and
D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the
D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these
particular model and would like to use them in an
Have you check the Selinux in Fedora Core is running ?
If yes , try to disable (need to reboot) , and try again.
I am facing some problem on webserver and NAT setting because of Selinux is
running.
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Sean Kennedy [EMAIL PROTECTED]
Cc:
uname -a
Hi,
I'm using latest Bristuffed Asterisk under Suse 9.2 and upgraded kernel.
Everything seems to be working fine, except crash when removing qozap with
modprobe -r qozap.
Any hint what's wrong ?
Thanks in advance,
regards,
Rob.
Linux voip 2.6.8-24.13-smp #1 SMP Fri Mar 18 10:19:42 UTC
I have my * test box connected to our phone system here on
an analog port with a X100P card.
The phone system requires a 9 to pickup an outside
line. Problem is, I think asterisk is passing the phone number to quickly
after the 9 and not getting the line. Is there any way to get
I do not have my config files as this computer but it seems to me you
are missing something on the end
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]/SOMETHING GOES HERE
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]/SOMETHING GOES HERE1
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]/SOMETHING GOES HERE 2
On Apr 6, 2005 2:29 PM, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Is there some sort of Syntax checker for the Asterisk config files. Also
is the /var/log/messages the only file that Asterisk dumps messages too
or is there another place. My system works but I want to make sure I am
not
I'm trying to get paging going on my [EMAIL PROTECTED] setup.
I was recommended the V-9970 because that
works with a FXO port, which I have one port open.Well I gotone the
other day and I can't seem to get it working. I wired it up
correctly because if I hook the phone jack onthere to an
J. Arnaud wrote:
Hi,
I am using the dial out feature
(/var/spool/asterisk/outgoing) but when I look in
CDRs,
calls that reached a all circuits are busy now,
please call later are considered as ANSWERED.
Is it the expected behavior? It there a way to change
that?
If you have analog calls are
I have searched high and low for these, but to no avail, nothing useful
back from google, nothing I could find on this mailing list, or
voip-user.org.
Does anyone have any good urls and or pointers which will assist in
configuring SIP Express Router and Asterisk talking to each other on the
same
Title: Call gets cut off after 5 minute
I had a call cut off after 5 minutes. I made the call through a SIP via * onto a x100p. 5 minutes into the conversation, the call got cut off. Is there a max time limit parameter somewhere in asterisk that I can change?
In my country payphone solutions for Call Shops are implemented using FXS
SIP or H.323 gateways that implement the Polarity reversal feature that
reverse polarity as soon as the other party answers. I have done this in
several VoIP platforms but Asterisk.
Regular Payphones and Call Shop metering
This may help, I just happen to be a google searching master :)
http://www.voip-info.org/wiki-Asterisk+at+large
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of G.Marshall
Sent: Wednesday, April 06, 2005 11:06 AM
To: asterisk-users@lists.digium.com
Subject:
I connect pri to asterisk with e100p card when i call from pri i cant hear
any sound And when i call ip phone icant hear any sound. Does any one have
idea
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Hello all,
I have noticed a few people have mentioned recently that they are
looking to set up BRI in the US. I am looking to do the same and I am
wondering if anyone has had any success yet? And, if so what card are
you using? I have heard that the Eicon Diva Server boards work, but
they are
Hi All,
I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit
of a problem. I can dial out with no problem but once * bridges the call to
another channel (such as a Zap channel to the PSTN or an internal Analog phone)
it appears that the keypad is then disabled which
Hi,
The problem is that when opening the zap channel, originate thinks
that the call has been answered and send the call to the beginning of
the context out. And what I really want is to make this but when the
destiny person answered and not when the zap channel opens.
as already in the
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?
This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command?
1)If you are usind Dial command, do not use T or t flags
2)DTMF mode
Thanks
But I was looking for a more complete solution like areski or astcc
I found nothing so I wrote my own, but they are for postgres. They are
not complete by no means. If you are interested, I will let you have a
look at what I have done, and if you provide constructive critisism, I
Marshall,
I am interested in seeing what you wrote to manage MySQL database
objects.
By the way, latest version of OpenOffice comes with a MySQL
Administrator GUI to manage tables and data. This is something to look
at too.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
I'm no Linux guru but my understanding is that the script you are referring
to (/etc/rc3.d/S40asterisk) just points to the real script in
/etc/rc.d/init.d or whereever (depends on which linux you're using). The rcn
(e.g. rc3) refers to the run level i.e. 3 in your case. So when you boot
linux
Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm
getting issues connecting between SIP phones.
A bit of background... I have an asterisk server running in a central location
where I have two incoming analog lines connected to FXO ports, two analog
I have a question regarding interoperability between Asterisk and an
Inter-Tel (I believe it's an Axxess) PBX at a client's location.
My clients currently have an Inter-Tel system that they're fairly happy
with but they're being bludgeoned by Inter-Tel for their proprietary IP
based phones.
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Email for details.
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You could try having a look at the CLI (asterisk
-rv) to ensure that Asterisk is doing what you think it is.
Regards
Cameron
- Original Message -
From:
Jeffrey
Sharpe
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, April 07, 2005
Hi...
Since I upgraded the asterisk version from 0.93 to
1.0.3 I have had a lot of problems to complete callsthru Vocal Sip Proxy
1.4.
When I just register Asterisk to receive calls from
vocal it works fine. Ex.
register =
2127701002:[EMAIL PROTECTED]/2098
And
When I just regiter
Hi,
I just got an Ingate Firewall, I cannot get asterisk to register to
it. Using UDP port forwarding to the Asterisk box, its like the
firewall is re-writing the password part.
I keep getting an error like the password is wrong, when trying to
route through the firewall.
In the console i get.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Stuart Hirst wrote:
Alessio,
I have also seen this problem on two different asterisk servers using
TDM400p cards.
I have not been able to resolve it. If you do an lspci you can see that
the system can see the devices but the zaptel drivers
hi
I created a .call file as mentioned in the WiKi but when i place it in
/var/spool/asterisk/outgoing, the Asterisk console shows unknown keyword for
all the keywords used in the .call file (i.e channel, context, extension,...).
Any ideas why?
Regards,
Gilbert Abboud
M.Eng. Computer
Let us try the opposite of what I have suggested and see what it does.
Change the Dial command as under and see how that goes.
exten = _9NXXNXX,1,Dial(Zap/2/${EXTEN:1},30, Tt)
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent:
Remove all the spaces in front of the lines
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gilbert
Abboud
Sent: Wednesday, April 06, 2005 3:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk .call files
hi
I created a
Hello,
I have implemented a test ACD with Asterisk 1.0.7, in which I have 2 agents and one user making calls and using AgentCallbackLogin. Besides that, I have other users on the PBX, but not necessarily members of any queue.
Agents and PBX users are using X-Pro as a soft-phone.
I am having
On April 6, 2005 03:47 pm, Gilbert Abboud wrote:
I created a .call file as mentioned in the WiKi but when i place it in
/var/spool/asterisk/outgoing, the Asterisk console shows unknown keyword
for all the keywords used in the .call file (i.e channel, context,
extension,...). Any ideas why?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi.
If you show what's in the .call file, it would help.
Rgs
Joao
Gilbert Abboud wrote:
| hi
|
| I created a .call file as mentioned in the WiKi but when i place it
| in /var/spool/asterisk/outgoing, the Asterisk console shows
| "unknown
-Original Message-
* is on a analog port of our legacy phone system which then converts it
to digital.
So do I use the W option?
-per your original post, on an x100 you would use the
Dial(zap/1/www${EXTEN})
-Jason Kawakami
___
No change whatsoever. I still think it's something with the phone... none of
the keys have any effect after dialling... they do not echo to the display
even. It's literally as though the keypad has been disabled.
Ian
[EMAIL PROTECTED] 06/04/2005 15:41
Let us try the opposite of what I have
Title: RE: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems
Well, the x100p is not always good either. If we forget
that it only support 600 ohm impedance, the proper example would bethe
problem i have and not being able to overcome is tremendous echo on the VOIP
phone when i
Hi,
I'm just curious if someone had/has a problem with livevoip. When I try
to make an outgoing call, I receive:
-- Called username:secret@217.160.244.186/x037378896
Apr 2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call
rejected by 217.160.244.186: No authority found
I have also seen this problem on two different asterisk servers using
TDM400p cards.
I have not been able to resolve it. If you do an lspci you can see that
the system can see the devices but the zaptel drivers don't see them.
I have other systems that work fine and so this has to
Run this from the CLI...
iax2 show registry
Do you see an entry that matches your LiveVoIP server IP (east or west
coast) and is it registered?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrejus
Stavickis
Sent: Wednesday, April 06, 2005 1:23
Stuart Hirst wrote:
Alessio,
I have also seen this problem on two different asterisk servers using
TDM400p cards.
I have not been able to resolve it. If you do an lspci you can see that
the system can see the devices but the zaptel drivers don't see them.
I have other systems that work
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