Hi,
On Tue, November 29, 2005 13:50 Francesco Peeters wrote:
BTW: BRIstuff is not included by default as it breaks PRI support.
Asterisk is already set up to use zap, so that is easy...
As far as I know, BRIstuff is not included for licencing reasons... Is
it true, that PRI support and
Dunno :)
what do you thing is wrong there? the compile was fine!
I only need a solution how to fix this error!!
On Sat, 03 Dec 2005 01:52:03 +0800
Steve Underwood [EMAIL PROTECTED] wrote:
How could a CVS update fix an error you have made during installation?
Steve
René Enskat [Teamware GmbH]
On Sat, December 3, 2005 9:28, Karsten Wemheuer said:
Hi,
On Tue, November 29, 2005 13:50 Francesco Peeters wrote:
BTW: BRIstuff is not included by default as it breaks PRI support.
Asterisk is already set up to use zap, so that is easy...
As far as I know, BRIstuff is not included for
I tried compiling Asterisk 1.20 (fresh cvs checkout of just some 15 minutes
ago) on a CentOS 4.2 box.
Compiling zaptel seems to work fine but when I try to compile * I get this:
chan_zap.c:8904: error: structure has no member named `useruserinfo'
chan_zap.c:8012: warning: unused variable
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by
Hi list:
Can any body gives me a voip provider with trunked
iax2 ,because i have tried voipjet and sixtel and they
are not trunked .
Regards;
chawki
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Hi I've been playing around with Realtime Asterisk using the ODBC
module to connect to my database and I got extensions working but now
I'm looking to get my iax.conf into the database. I would like to have
the users who can register with my box to dial extensions in there,
and also the
jonny hashem a écrit :
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
--
Hi:
Now the time out is message is gone ,why the call
still fails?
--- Administrator TOOTAI [EMAIL PROTECTED] wrote:
jonny hashem a écrit :
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
dial [EMAIL PROTECTED]
Just a quick note to say thanks to all who replied, most helpful.
Thanks Again
Scott
scott wrote:
Hi All
I am using prepaid auth (callingcards), the idea is for a prepaid support line.
It is up and running but I have a couple of questions with regards to
modifications I would like to make.
chawki hammoud a écrit :
Hi:
Now the time out is message is gone ,why the call
still fails?
Do an iax2 debug, set verbose 5 and check in logs.
--
Daniel
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On Fri, Dec 02, 2005 at 08:24:58PM -0500, Jerry Geis wrote:
hcitool cc MACHEADSET
hcitool auth MACHEADSET
hcitool dc MACHEADSET
rfcomm bind rfcomm0 MACHEADSET
sdptool search --bdaddr MACHEADSET 0x111E
These Steps are not necessary, since chan_bluetooth does this for you.
however you
On Saturday 03 December 2005 04:09, Remco Barendse wrote:
chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug'
chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this
chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from
chan_zap.c:10871:
On Sat, 3 Dec 2005, Andrew Kohlsmith wrote:
On Saturday 03 December 2005 04:09, Remco Barendse wrote:
chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug'
chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this
chan_zap.c:10870: warning: passing arg 1 of
Hello,I am looking to hookup my Asterisk box to a gateway provider. I want the cheapest possible rates with highest reliabilty. Countries I am looking for are 1. Pakistan. 2. India. 3. Hong Kong 4. Singapore 5. ChinaIt does not need to be the same provider for all countries. Like I
On Saturday 03 December 2005 04:09, Remco Barendse wrote:
chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug'
chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this
chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from
Hi,
does anyone has experiance with connecting a PRI line to a TE405P card
together with zaptel 1.2.0?
We are located in Germany and there is nos CID number for incoming calls.
What are the right settings for zaptel.conf and zapata.conf?
Thanks and regards,
bk
I had the same problem, I am using Asterisk 1.2.
I installed spandsp 0.0.3pre6 on the machine then I installed
spandsp-0.0.2pre21c I received the same error
fax_set_phase_d_handler. It seems I didn't get all of the files
associated with 0.0.3pre6 removed before I installed 0.0.2pre21c.
To
Hi everyone !
I'm testing Queue Statistics 0.6 from AsteriskGuru Tools with [EMAIL PROTECTED] 2.0, and i gota problem; every call is registered as a new agent when i have configured Static Agents in a Queue, because of Local/[EMAIL PROTECTED]
I changed to dynamic agents, and it worked only for
On Sat, 3 Dec 2005, Rich Adamson wrote:
On Saturday 03 December 2005 04:09, Remco Barendse wrote:
chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug'
chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this
chan_zap.c:10870: warning: passing arg 1 of
On Fri, December 2, 2005 22:54, Francesco Peeters said:
On Fri, December 2, 2005 22:50, Francesco Peeters said:
On Fri, December 2, 2005 21:45, Kristof Hardy said:
Francesco Peeters wrote:
Does anybody have any experience in this?
I am using * 1.2 BRIstuffed 0.3.0 Pre1
No experience on
I just upgraded my config from * 1.0.10 to 1.2
I removed caller ID from my configs because when I try to use CallerID
(new style) on my IAX provider (magrathea) but whenever I try to make a
call I get a message from the provider that You are not registered to use
this service. Removing the
On Dec 2, 2005, at 12:03 PM, Jess Coburn wrote:
Guys,
I'm curious if it's possible to asterisk at home and the sangoma T1
cards together. I realize asteriskathome is traditionally used for
at home, but I'd like to use it in a small office with a T1 and our
hardware is a Sangoma card. I
I have had the same issue. The headset connects, but there is no audio.
I have a IOGEar usb dongle. However, if i use my Q-Stor usb bluetooth
dongle I do have audio.
Doing a sdptool browse for the rfcomm channel. it is the same in both cases.
So it's something with the dongle not accepting some
On Sat, 3 Dec 2005, Remco Barende wrote:
Whenever I pick up that phone I get on the console:
Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel
'Zap/1-1' sent into invalid extension 's' in context 'default', but no
invalid handler -- Hungup 'Zap/1-1'
Have
Hello again
i'm very new with this theme
i'm testing the [EMAIL PROTECTED] 2.1 recently i donwloaded from the
sourceforge.net
i need info about the intel chipset modem to call and receiving calls
and the configruration for internal extensionsl work ok, i install in 3
computers the xten free
Thanks for the feedback guys. yeah I RTFM'd last time but honestly wanted the box up quickly so I gave up too early. Today, I'm reloading it now. I have the 2.1 ISO installing but I read in their forums there's possibly a problem with the install that's on the Georgia mirror and that the UK mirror
On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said:
On Fri, December 2, 2005 22:54, Francesco Peeters said:
Watching the console for a while I see regular messages, which I could
also find in /var/log/messages:
Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel
On Sat, 3 Dec 2005, Begumisa Gerald M wrote:
On Sat, 3 Dec 2005, Remco Barende wrote:
Whenever I pick up that phone I get on the console:
Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel
'Zap/1-1' sent into invalid extension 's' in context 'default', but no
On Sat, 3 Dec 2005, Francesco Peeters (Asterisk) wrote:
On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said:
On Fri, December 2, 2005 22:54, Francesco Peeters said:
Watching the console for a while I see regular messages, which I could
also find in /var/log/messages:
Dec 3
On Sat, 3 Dec 2005, Remco Barende wrote:
I have but only for the phone line, it is immediately after:
signalling=fxs_ks
immediate=yes
What I actually meant is that you should turn this off if you don't need
the functionality. Most likely you are defining the extension
I experienced a similar situation with the SPA-841, it turned out to be
that the calls I was missing didn't have caller ID (outside calls with
caller ID Blocked), found that the SPA841 phone has an option to ignore
calls without caller ID. Turned this option off and it fixed the
problem.
Sorry, I
Hi,
Can I escape a call queue by pressing a '*' or do I have to use a digit??
Thanks
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in wath link or page is the * commands for the phone extensions??
example *79 is for on or off the extension
??
Thanks again in advance
- Original Message -
From: Chuck Bunn [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, December 03, 2005 2:06 PM
Subject:
Im in the same situation, I have been trying to google around a lot for some examples but havent come up with much.
Hi I've been playing around with Realtime Asterisk using the ODBCmodule to connect to my database and I got extensions working but nowI'm looking to get my iax.conf into the
I'm getting much nearer in getting my Sangoma A2022-SO analog card working
with Asterisk 1.2, however I am unsure of the ordering of ports on the
rear of the card.
I've taken some pictures of the card in the hope anyone can help me
guess which physical ports relate to the 2 x fxs and 2 x fxo
Hi:
i made the debug and look what i get:
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: NEW
Timestamp: 00016ms SCall:
*411 Directory
*43 Echo Test
*60 Time
*61 Weather
*62 Schedule wakeup call
*65 festival test (your extension is XXX)
*70 Activate Call Waiting (deactivated by default)
*71 Deactivate Call Waiting
*72 Call Forwarding System
*73 Disable Call Forwarding
*77 IVR Recording
*78 Enable Do-Not-Disturb
what is your question? you must set up extensions yourself in
extensions.conf...you can set the extensions to whatever you want
(say, if you are replacing an existing PBX and want the users to have
the same extensions).
-yair
On 12/3/05, Vladimir Montealegre [EMAIL PROTECTED] wrote:
*411
Very slow response, if they have the DID
you want in stock you get it right away, they dont seem to be able to
port toll or DIDs as advertised. They claim to have a server in Washington State
and Florida
but no centralized server, they are too far for me to consider good service.
On Sat, December 3, 2005 19:01, Remco Barende said:
On Sat, 3 Dec 2005, Francesco Peeters (Asterisk) wrote:
On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said:
On Fri, December 2, 2005 22:54, Francesco Peeters said:
Watching the console for a while I see regular messages,
On 3 Dec 2005, at 20:27, chawki hammoud wrote:Hi:i made the debug and look what i get: dial [EMAIL PROTECTED] -- Executing Dial("OSS/dsp","iax2/callshopcompany/0017046872001") in new stack -- Called callshopcompany/0017046872001Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:IAX
Hi:
Thanks for your answer, i tried all possible codecs
and the same result the call failed,my asterisk
verison is 1.0 ,I asked callshopcompany the voip
provider about whats the reason of the failure of the
calls and he said he didnt know whats the problem and
he's all customers making succesful
-- Called 1101
-- Agent/1101 is ringing
-- Got SIP response 480 Temporarily Unavailable back from x.x.x.x
-- SIP/1101-9b08 is circuit-busy
Is it possible to force logoff such agents?
--
Vladimir S. Blazhkun, Personal Communications Systems, LLC.
Leading IP NCC Specialist,
Hi,
I understand that I must set up extensions myself but in the release
notes for Asterisk 1.2 it specifically states that multiple digit
extensions can be used in the exit context of a queue. Prior to version
1.2 only a single digit would would when exiting from a queue. I have
tried
I just wanted to say thanks to everything that provided feedback and assistance on this. I have AAH2.1 running with my Sangoma T1 card. Here's a few of the gotchas I ran into and my process, please note I'm not an asterisk expert and this could be totally wrong but here's what worked for me:
1.
Doug Lytle wrote:
Hey everybody.
I have an Adit 600 that I'm not able to get working properly with
Groundstart. The Adit (BootCode 1.23), 1 TDM (Version 6.1.2) and 1
FXO card (Version 1.12).
For the sake of completeness and the archives, the answer to this
problem was discovered in
I have similar problems with call drops.I don't know if "Shadow Ping" is some kind of pinging software. I have run a lot of flood pinging and everything comes back just fine. I don't have Cisco phones, I use Softphones and it's the only application running on the PCs (aside from MS Windows and
Hi,
I setup music on hold as directed for Asterisk 1.2 but still no music on
hold. Any ideas what I did wrong. I see it start in the CLI but then it
immediately stops?? I also see the Hangup occur 20 seconds later as it
should according to WitMusicOnHold(20). I used a test setup suggested in
I had the same problem, had to reboot the machine it started to work.
ThanksOn 12/3/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,I setup music on hold as directed for Asterisk 1.2 but still no music onhold. Any ideas what I did wrong. I see it start in the CLI but then itimmediately stops?? I also
I have actually spoken to a Polycom product engineer at VON and brought
this up. It seems like it's a frequent request, hopefully they will
address it soon.
On 12/2/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Official Polycom view seems to be that you shouldn't work at night :)The phones are
This is correct, astdb isn't stored in a database and essentially your
scenario with astdb file being shared won't work or will be highly
unreliable. I am almost sure there was a patch in circulation, alas I
didn't find it in the bug tracker.
On 12/1/05, Bruce Ferrell [EMAIL PROTECTED] wrote:
Matt
4) What is some good company
names to purchase
DID's and VoIP termination from? I have been looking at VoIPJet and
Teliax.
In my six months of using both
services full time, both have worked reliably, but we use Teliax for
DIDs as we need a company we can talk to when we need something,
I had deployed asterisk on my Toshiba Satellite laptop and it was
running successfully, later I tried migrating to postgreSQL database,
towards that i deployed the database server also on the same laptop
and tried to configure the files as required. Somewhere i went wrong
and now
hrishikesh shrivastaw wrote:
[cdr_pgsql.so]Dec 4 11:56:02 WARNING[3839]: loader.c:258
ast_load_resource: libpq.so.4: cannot open shared object file: No such
file or directory
Dec 4 11:56:02 WARNING[3839]: loader.c:440 load_modules: Loading
module cdr_pgsql.so failed!
Check that
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