Hi all,
Just want to check from the list experienced personal about the Digium
HPEC, where I had purchased the HPEC and wish to run with TDM card
Sangoma A200. I can't install HPEC to run with Sangama A200 card, even I
had changed my hpec file from i686 to i386.
The error that I had as bellow;
Am Sonntag, den 06.01.2008, 21:05 -0600 schrieb Alejandro Kauffmann:
As long as this is an official rant thread
Good to know no new phones have hit the market since the last time this
question was asked and answered. It's also good to know
opinions about specific products don't change
Hi guys,
Can someone tell me if there is a way to increase the volume of a conversation
that occurs between two SIP channels or between a SIP and an IAX channel ?
My headsets are set to the maximum volume but the voice is still low ... I know
there is a configuration in zapata.conf for the
[EMAIL PROTECTED] wrote:
Can someone tell me if there is a way to increase the volume of a
conversation that occurs between two SIP channels or between a SIP and an IAX
channel ?
My headsets are set to the maximum volume but the voice is still low ... I
know there is a configuration in
I do have to answer to your suggestion of renaming the CTLSEPmac.tlv
to SEPmac. The phone is still requesting CTLSEPmac.tlv and as it
cannot find that it goes into a loop. I also let the phone do that the
whole weekend so there should be no iterative process in requesting the
files as I read in
Philipp Kempgen wrote:
Bhrugu Mehta wrote:
I am new to zaptel programming.
can anybody help me how to start this. or any ref. site or matirial
availabel.
i want to use c lang. for this.
sarcasm mode=SCNR class=ignore
Some tutorials:
http://www.google.com/search?q=learn+c+in+21+days
Hi all!
Sorry for my poor english, i'm italian.
I installed Digium B410P on my asterisk server. I followed the official
installation instructions found on digium site. These instruction, in my
opinion, are not clear, so i tried with other ways (found on the trixbox
site). I found this:
Ditto !!!
- Original Message -
From: Bill Hackensack
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, January 06, 2008 11:14 PM
Subject: Re: [asterisk-users] Which IP Phone is really the best?
Wow! That is a good question. I can't believe no one
Update and revision:
I now downloaded the oldest gettable SIP firmware for 7941/61, i.e.
8.0.2. I always get the same behaviour. But I realized it never got to
the SIP image completely loaded status.
I bought this phone and it had - no wonder - an SCCP image installed.
When plugging that into an
On Mon, Jan 07, 2008 at 12:36:16PM +0530, Bhrugu Mehta wrote:
hi, all
I am new to zaptel programming.
can anybody help me how to start this. or any ref. site or matirial availabel.
i want to use c lang. for this.
thnks in advance.
Apart from the obvious reference to the source itself
Hi All,
I have a problem with incoming calls to Asterisk that go to SIP phones, when
the h.323 message contains:
..11 = Screening indicator: Network-provided (0x03)
.01. = Presentation indicator: Presentation restricted (0x01)
for the calling party number, the Contact URI
Hi Daniele,
Could you please tell us what exactly happens?
Are your able to see some error in the log/console?
On Jan 7, 2008 11:53 AM, daniele visaggio [EMAIL PROTECTED]
wrote:
Hi all!
Sorry for my poor english, i'm italian.
I installed Digium B410P on my asterisk server. I followed the
Hello,
I have the following problem. I am migrating my asterisk infrastructure
to a new server and I encounter a strange problem. The configuration is
as followin: IAX clients connect to asterisk which forward calls to a
sip box connected to a phone line. On the old server everything works ok
but
Maybe it was all compiled out of order.
I believe that Zaptel has to be compiled AND installed before compiling
asterisk to compile the zap channel.
--
--
Steven
http://www.connectech.org/
Raúl Gómez C. [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Well, problem solved,
I am using freePBX, so my dialplan uses macros and such, but here is what I do.
exten = 91248640ABCD,1,Goto(outrt-006-CellGateway,394${EXTEN},1)
;I have a list of all of our company's cell phone numbers. (We get free Cell to
Cell)
[outrt-006-CellGateway]
include = outrt-006-CellGateway-custom
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
(if you set in /etc/default/zaptel: ZAPBRI_SIGNALLING=bri
you'll get that from genzaptelconf.
If I create a file like this, I end up with signalling=bri_cpe instead of
signalling=bri_cpe_ptmp.
Anyway, either you use zaphfc or vzaphfc. The first one
Hi,
I am trying to connect the outbound dialing with mysql with the following
code,
exten = 88,1,MYSQL(Connect connid hostname username password dbname)
exten = 88,2,GotoIf($[${connid} = ]?error,1)
exten = 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\
tablename\ WHERE\ phone =${a})
Gopal krishnan wrote:
Hi,
I am trying to connect the outbound dialing with mysql with the
following code,
exten = 88,1,MYSQL(Connect connid hostname username password dbname)
exten = 88,2,GotoIf($[${connid} = ]?error,1)
exten = 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\
clive.chan(Atn) wrote:
The error that I had as bellow;
Found key 'HPEC-XX' for 2 channels.
Found valid HPEC licenses for 2 channels.
Failed to get license challenge: No such device
Please contact Digium support for assistance with using HPEC since you
purchased it.
--
Kevin P.
Lee Jenkins wrote:
Philipp Kempgen wrote:
Bhrugu Mehta wrote:
I am new to zaptel programming.
can anybody help me how to start this. or any ref. site or matirial
availabel.
i want to use c lang. for this.
sarcasm mode=SCNR class=ignore
Some tutorials:
Hi Daniele,
Could you please tell us what exactly happens?
Are your able to see some error in the log/console?
Thanks for your answer.
I'm managing the asterisk server from a windows client via ssh (putty
client), so i can't paste here the output of the asterisk CLI, but when i
try to do
daniele visaggio wrote:
Thanks for your answer.
I'm managing the asterisk server from a windows client via ssh (putty
client), so i can't paste here the output of the asterisk CLI, but
when i try to do a call to the PSTN i see a lot o f messages, but
Putty does support copying all the
Daniele,
you need an external calls rule in your extension.conf, that is 1 to call
using PSTN line.
Please send your extension and we can take a look to find your problem.
p.s.
I'm Italian too.
On Jan 7, 2008 3:03 PM, daniele visaggio [EMAIL PROTECTED] wrote:
Hi Daniele,
Could you please
On Mon, Jan 07, 2008 at 03:03:18PM +0100, daniele visaggio wrote:
I'm managing the asterisk server from a windows client via ssh (putty
client), so i can't paste here the output of the asterisk CLI,
Huh???
* You can record history in putty.
* You can manually copy text from putty( just mark
Hi,
I feel I've read a thread about this previously but I couldn't find it.
Is there way for an Asterisk server to check if a sip phone is forwarded
without bothering phone's user ?
I was thinking of some Alert-Info option that would let the phone reply with
a 302 Moved Temporarily or 182 Queued
Olivier wrote:
Is there way for an Asterisk server to check if a sip phone is forwarded
without bothering phone's user ?
No.
I was thinking of some Alert-Info option that would let the phone reply
with a 302 Moved Temporarily or 182 Queued message and not let the phone
ring or display
2008/1/7, map [EMAIL PROTECTED]:
Daniele,
you need an external calls rule in your extension.conf, that is 1 to
call using PSTN line.
Please send your extension and we can take a look to find your problem.
p.s.
I'm Italian too.
Ok, i attach my extension.conf.
Thank you very much, i'm
Hello All,
I have a problem. I have tried everything that is in the book The
Future of Telephony as well as on the FWD (freeworlddialup) website,
and there is still a problem. My asterisk box is not able to associate
with the FWD server. I get:
Registration Rejected by [insert IP], and I can't
Tzafrir Cohen wrote:
On Mon, Jan 07, 2008 at 03:03:18PM +0100, daniele visaggio wrote:
I'm managing the asterisk server from a windows client via ssh (putty
client), so i can't paste here the output of the asterisk CLI,
Huh???
* You can record history in putty.
* You can
I have a TDM400 in the server. I want to press **1XX to pickup a
call. It is ok if I pickup a call dialled from 1XX to 1YY (internal
network call). However, it is failed to pick up a call from PSTN
thro' TDM400 card. It seems I can't guess the correct context of it.
How can I know the context
On Mon, Jan 07, 2008 at 10:19:11AM -0500, John Novack wrote:
The ONLY issue I have had with PuTTY is ( my ) inability to run make
menuselect. regardless of how I set PuTTY, it complains about terminal size.
Hmm.. I have never encountered this. Sounds like a bug. Can you point me
to a bug
2008/1/7, map [EMAIL PROTECTED]:
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put a
number greater than 200 :-). I suggest 10.
Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the
strike out stable on the cisco phones. they are not stable.
On 1/7/08, Tim Connolly [EMAIL PROTECTED] wrote:
Cisco 7960's: (SIPified)
1. Cheap
2. 6 lines is plenty
3. simple to config
4. stable
On Jan 6, 2008, at 11:03 PM, William Herrera wrote:
Alright, enough.
At first I was to
Hi,
Can someone point me to a zapata.conf example that will create a
single DIAL OUT
group including all 4 spans on a TE4XXP?
One friend says to change the group number all to 1 on all 4 spans.
Another suggestions says it is possible to have these unique groups (1-4)
and to combine all 4 into a
Hello all,
Can anyone have any experience working with service provider
like Talkfree .
They are giving the user accounts based on the single user accounts
and those needs to be directly register to the service provider not to the
local system
i have taken a connection which when configured
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put a number
greater than 200 :-). I suggest 10.
On Jan 7, 2008 4:00 PM, daniele visaggio [EMAIL PROTECTED] wrote:
2008/1/7, map [EMAIL PROTECTED]:
Daniele,
you
On Mon, 2008-01-07 at 12:15 +0100, Christophorus Laube wrote:
Update and revision:
I now downloaded the oldest gettable SIP firmware for 7941/61, i.e.
8.0.2. I always get the same behaviour. But I realized it never got to
the SIP image completely loaded status.
I bought this phone and it had
You haven't said if your connection to fwd is SIP or IAX2 but I have
found IAX2 connections to fwd to be unreliable. Other people may have
different results.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 10:17
To:
It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?
If so, how would I go about that.
On 1/7/08, Huw Richards [EMAIL PROTECTED] wrote:
You haven't said if your connection to fwd is SIP or IAX2 but I have
found IAX2 connections to fwd to be unreliable.
On Mon, 2008-01-07 at 01:50 +0200, Tzafrir Cohen wrote:
On Mon, Jan 07, 2008 at 12:41:11AM +0100, Hans Witvliet wrote:
On Sat, 2008-01-05 at 13:36 -0500, Shane D wrote:
Hello List,
I am getting Asterisk set up. I am going to be installing Debian Linux
on a laptop later. I would
I have a problem. I have tried everything that is in the book The
Future of Telephony as well as on the FWD (freeworlddialup) website,
and there is still a problem. My asterisk box is not able to associate
with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use
I will try that when I return home in about two hours. I'll let you know.
On 1/7/08, Benchev [EMAIL PROTECTED] wrote:
I have a problem. I have tried everything that is in the book The
Future of Telephony as well as on the FWD (freeworlddialup) website,
and there is still a problem. My
Hi,
Asterisk now supports h234m.
Does anyone know a Media gateway such as those of Mediatrix, Patton,
Audiocodes, Cisco that also supports h324m flows ?
Prospective setup would be:
ISDN --PRI-- Media gateway --SIP -- Asterisk ---SIP --- Videosoftphone
Cheers
Hi
On an old IBM Netvista thinclient, the TDM card doesn't detect
incoming calls, although the card seems to be detected, and correctly
configured:
pbx asterisk # lspci
00:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
# cat /etc/zaptel.conf
fxsks=1
Hi,
I'm probably missing something trivial but I don't
understand what.
Asterisk is loading fine but when I connect to the
console (asterisk -vr) and type stop I get a no such
command reply:
*CLI help
(...)
skinny show lines Show defined Skinny lines
per device
soft
Vieri:
You will need to specify to the stop command, when to stop. The options are:
now, when convenient or gracefully.
Running the command 'help stop' inside the CLI will give an idea of this
options.
Best regards, Tomás.
On Jan 7, 2008 5:21 PM, Vieri [EMAIL PROTECTED] wrote:
Hi,
I'm
On Jan 7, 2008, at 8:53 AM, daniele visaggio wrote:
2008/1/7, map [EMAIL PROTECTED]:
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put
a number greater than 200 :-). I suggest 10.
Sorry, i'm using a Linux
Hi
Nobody has an Idea ? Should I try and fill a bug report (or feature
request ?) at Digium ?
The only solution I personally see is a patch in the source.
Regards
Gaetan
On 04/01/2008, at 23:26, Gaëtan Minet wrote:
Hi everybody
We have a strange problem with several asterisk servers
Greetings all,
I'm not real good with dial plan programming and need some help. I've looked
at the 2nd edition of the Asterisk book about GotoIf() and have a basic idea
what I need to do but not sure about the correct way or the best way, to set
it up. I need to branch based on whether the
So, watching the asterisk console with full debug on shows something
about Starting Music On Hold for Channel xx/yy-zz?
Shane
On Jan 7, 2008 11:00 AM, Gaëtan Minet [EMAIL PROTECTED] wrote:
Hi
Nobody has an Idea ? Should I try and fill a bug report (or feature request
?) at Digium ?
The
Mike Trest - Personal wrote:
Hi,
Can someone point me to a zapata.conf example that will create a
single DIAL OUT
group including all 4 spans on a TE4XXP?
One friend says to change the group number all to 1 on all 4 spans.
Another suggestions says it is possible to have these unique groups
--- Tomás Laureano Peralta Tormey
[EMAIL PROTECTED] wrote:
Vieri:
You will need to specify to the stop command, when
to stop. The options are:
now, when convenient or gracefully.
Running the command 'help stop' inside the CLI will
give an idea of this
options.
Thanks, I know that 'stop'
Christophorus Laube wrote:
I do have to answer to your suggestion of renaming the CTLSEPmac.tlv
to SEPmac. The phone is still requesting CTLSEPmac.tlv and as it
cannot find that it goes into a loop.
Did you create a zero byte file with that name?
I had to do that with a 7960 and it was very
I get the following output:
Jan 7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
Registration of '886036' rejected: 'Registration Refused' from:
'192.246.69.186'
It shouldn't be trying to do something on my network (192) should it?
On 1/7/08, Shane D [EMAIL PROTECTED] wrote:
I will try
Ignore the part about 192... That happens to be the IP of iax2.fwdnet.net.
If anyone has set up FWD successfully for incoming only, let me know.
On 1/7/08, Shane D [EMAIL PROTECTED] wrote:
I get the following output:
Jan 7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
Registration of
Although it's not LDAP I used a script that I found on the voip wiki and
changed it so it looked at only sip configuration files. It also
alphabetizes the output so it can be displayed that way on the phone.
Below are my notes on the subject. If someone is willing to post this
to the wiki and
192.246.x.x =! 192.168.x.x if that is what you're thinking.
Anyway, registration Refused sounds like you're getting up to FWD and
attempting to authenticate, but failing at that point.
double-check your iax.conf settings against the FWD Extra Features
settings... of course make sure IAX is
Vieri wrote:
Hi,
I'm probably missing something trivial but I don't
understand what.
Asterisk is loading fine but when I connect to the
console (asterisk -vr) and type stop I get a no such
command reply:
*CLI help
(...)
skinny show lines Show defined Skinny lines per
My config is as follows
Excerpt of sip.conf:
[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833
register = fwd-id:fwd-pwd@fwd.pulver.com/fwd-id
[sipfwd]
type=peer
secret=fwd-pwd
username=fwd-id
Try using the IP address, and not the dynamic URL, does anything change?
IPKall
IPKall Forum
http://voxilla.com/PNphpBB2-viewforum-f-38.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Huw Richards
Sent: Monday, January 07, 2008 1:55 PM
To:
Thank you for your reply Trevor.
Is there an easy way to achieve this with a computer generated voice? We do not
wish to manually record the messages if possible, in the interests of a
consistent message across all voicemail boxes. What would be the easiest way to
do this?
Also, can you
Hi Marc;
Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.
Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are
Hi Marc;
Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.
Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are
Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?
I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall to
the server directly, I don't need FWD...
On 1/7/08, Huw Richards
--- MatsK [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi,
I'm probably missing something trivial but I don't
understand what.
Asterisk is loading fine but when I connect to the
console (asterisk -vr) and type stop I get a no
such
command reply:
*CLI help
(...)
Still rejected.
On 1/7/08, www.IPKall.com [EMAIL PROTECTED] wrote:
Try using the IP address, and not the dynamic URL, does anything change?
IPKall
IPKall Forum
http://voxilla.com/PNphpBB2-viewforum-f-38.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Greetings!
We have a somewhat noisy background in our call center, and I'd like
to reduce this. Obviously, we could plaster the walls with sound
absorbing material, but is there anything we can do in software
either using any algorithms for our open source-based SIP library or
inside
My question would actually be - is there any support for h234 over ISDN?
PaulH
On Mon, 2008-01-07 at 19:59 +0100, Olivier wrote:
Hi,
Asterisk now supports h234m.
Does anyone know a Media gateway such as those of Mediatrix, Patton,
Audiocodes, Cisco that also supports h324m flows ?
Hi All,
kind of need help on the conference module, i'm using freepbx and
enabled conferencing, i created a conference number, 6000. when i dial
to it, my phone says it is connected but i'm hearing nothing, maybe logs
below can help you.
also, when i hang up the phone, the conference did not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Norman Franke wrote:
Greetings!
We have a somewhat noisy background in our call center, and I'd like to
reduce this. Obviously, we could plaster the walls with sound absorbing
material, but is there anything we can do in software either using
If you want to forward your ipkall number directly to your asterisk
server:
1. If your asterisk server is on a private LAN and is connected to the
internet via a router, enable the router to port forward UDP/5060
UDP/1-2 to your asterisk server (assuming you have not changed
rtp config
Hi,
I'm trying to use a mobile phone (ericsson W300i) with asterisk through
bluetooth. After some sutrggling, I foun chan_mobile.
As some one already used this mobile with what result?
I'm considering a simple asterisk system (for home use/test purpose) with :
- one SIP service provider (with
Wouldn't you need someone besides yourself in the conference?
On 1/7/08, Nhadie [EMAIL PROTECTED] wrote:
Hi All,
kind of need help on the conference module, i'm using freepbx and
enabled conferencing, i created a conference number, 6000. when i dial
to it, my phone says it is connected
no-ip.org appears to want to charge me money... Is there a free alternative?
On 1/7/08, Huw Richards [EMAIL PROTECTED] wrote:
If you want to forward your ipkall number directly to your asterisk
server:
1. If your asterisk server is on a private LAN and is connected to the
internet via a
http://www.no-ip.com/services/managed_dns/free_dynamic_dns.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 19:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and
Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about Use your IPKall number as the sip number I am
signing up for IPKall... Right?
On 1/7/08, Shane D [EMAIL PROTECTED] wrote:
no-ip.org appears to want to charge me money... Is there a free alternative?
On
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Shane D wrote:
no-ip.org appears to want to charge me money... Is there a free alternative?
Dyndns.org
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Paul Hales wrote:
My question would actually be - is there any support for h234 over ISDN?
Yep, but best place to ask about it is the asterisk-video mailing list.
You'll probably want to check out the work over at
sip.fontventa.com
- --
Kind
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Shane D wrote:
Wouldn't you need someone besides yourself in the conference?
Indeed, judging by the logs (last line) you are actually in a
conference, you'll need to get someone else to call the same number to
be able to talk to them.
Alternatively
hi shane,
thanks for your reply. i actually tried 3 phones dialled to the
conference, but cant here anything from those phones. i also enabled the
usercount so i can hear something at least. but still no sound.
i'm using ztdummy, as i dont have a card yet.
regards,
nhadie
Shane D wrote:
I did everything, and when I dial, nothing comes up in the consol,
nothing rings, and the phone says I'm sorry, but the person you are
trying to call has a mailbox that has not been configured yet. Good
bye.
What's wrong?
On 1/7/08, Shane D [EMAIL PROTECTED] wrote:
Okay. What do you mean in
I think you said that you already had an ipkall account? If so, logon to
the account and on the resulting screen there are 2 fields that you need
to change:
The first is the SIP Phone Number - if you have already tried to
forward the IPKALL number to your fwd account, this field will contain
your
--- Vieri [EMAIL PROTECTED] wrote:
*CLI stop
No such command 'stop' (type 'help' for help)
There was a config error on my behalf in the zapata
config and that somehow didn't stop asterisk from
loading but without the stop and zap commands.
Solved.
I'm an idiot... I dialled wrong on my phone... I changed it, and was
able to use the Echo application. Dialling for a call to my softphone
as we speak!
On 1/7/08, Huw Richards [EMAIL PROTECTED] wrote:
I think you said that you already had an ipkall account? If so, logon to
the account and on
You're right that dyndns.org offers the same type of services as
no-ip.org.
However, when I first setup ipkall forwarding directly to my asterisk
server (about a year ago), it would not work with a dyndns.org account -
I forget the reason why. Maybe ipkall works with dyndns now - I haven't
tried
Below is what I got from CLI
[Jan 7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 111.
On Jan 7, 2008 11:48 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
I have a TDM400 in the server. I want to press **1XX to pickup a
call. It is ok if I pickup a
It works! It works! It works!
I am able to talk to myself. Now all I have to do is write my dialplan...
Say, would I have to use the [ipkallnumber] extention? Could I specify
the s extension instead to catch multiple numbers?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Huw Richards wrote:
You're right that dyndns.org offers the same type of services as
no-ip.org.
However, when I first setup ipkall forwarding directly to my asterisk
server (about a year ago), it would not work with a dyndns.org account -
I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nhadie wrote:
hi shane,
thanks for your reply. i actually tried 3 phones dialled to the
conference, but cant here anything from those phones. i also enabled the
usercount so i can hear something at least. but still no sound.
i'm using
I seem to recall the problem was related to DNS A vs SRV records. I
believe that dyndns at that time did not register SRV records on host
(i.e. free) accounts and ipkall was looking for an SRV record.
I know that an SRV record can be added on a paid account, but I still
don't think that you can
You would have to try out the s extension yourself.
I tend to have different contexts for each incoming number (as a home
user, only one number at a provider) so I can potentially handle them
differently i.e. time of day check.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
daniele visaggio wrote:
2008/1/7, map [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put
a number greater than 200 :-). I suggest 10.
Sorry, i'm
Oh. Well, I want to design a dialplan, and I don't care what number it
is, as long as you get my starting menu... So I'll try out the S
extension.
On 1/7/08, Huw Richards [EMAIL PROTECTED] wrote:
You would have to try out the s extension yourself.
I tend to have different contexts for each
On Mon, Jan 07, 2008 at 04:53:03PM +0100, daniele visaggio wrote:
2008/1/7, map [EMAIL PROTECTED]:
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put a
number greater than 200 :-). I suggest 10.
Hi All
We received following error .Please help us to sort out.
WARNING[3281]: frame.c:1426 speex_samples: Had error while reading wideband
frames for speex samples.
Regards
Nirukshitha
Hi Matt,
it seems i don't have that command.
*CLI zap show channels
No such command 'zap' (type 'help' for help)
*CLI
! abort add ael agent agi
cdr databasedebug dnsmgr dontdump
dundi
extensions feature group help
Try 'ip4000_1' instead of '207' for your address.
CP
Kevin DeGraaf wrote:
I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on
a flat local network.
I followed the provisioning guides that I found on the Web, and I have
the phone downloading bootrom.ld, sip.ld, and a
Hi All,
Where can I find copies of the app_rxfax.c, app_txfax.c and
apps_Makefile.patch. They don't seem to be located at soft-switch.org
anymore.
I am currently trying to compile Asterisk 1.2.26.1 and need the fax
components.
Thanks.
___
Well, what would you have one say?
It is caused by the return of a failure value of the function
speex_get_wb_sz_at() in frame.c, which attempts to extract bits
from speex frames. Presumably from some sort of data corruption or
invalid format.
On Mon, 7 Jan 2008, pgck nirukshitha wrote:
Hi
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nhadie wrote:
Hi Matt,
it seems i don't have that command.
:)
You'll need to make sure that:
1. You have zaptel compiled
2. You compile Asterisk *after* zaptel is compiled and installed
3. You have either modprobed zaptel + ztdummy or made the
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