Hi
I would suggest a IAX2 trunk between the two servers. You will need to modify
the dialplan to recognise which extensions are on each box and route
accordingly. The fact your clients are SIP does not preclude you from using
IAX2 to connect the servers.
Regards
Jon
Jon Farmer
Telford,
Dear all,
now we have the same problem of res_perl compilation with asterisk 1.4. It is
the same problem that was present when asterisk was upgraded to version 1.2.
I hope Anthony Minessale will be able to solve that problem as he did on that
case. But if any of you know a hack to this
On Thu, Dec 28, 2006 at 12:35:57PM +1300, Ray Jackson wrote:
Dan,
I have IMAP support working now with Courier IMAP. Since Courier (and
probably Dovecot) do not support a single authuser connection that may
access any mailbox, you have to omit the 'authuser' and 'authpassword'
settings
Hi,
Well, in our case, it seems that the issue was being caused by
announcements. That is, someone in QUEUE1 would be waiting 15
minutes.. and QUEUE2 would be waiting 5 minutes. The person in
QUEUE1 would be listening to 'we're sorry you are holding so long, if
you'd like to leave a message,
Of course everyone is allowed to use VoIP... Asterisk is open! I think
Dovid's point was more that this guy's website says he buys and sells
precious metals and other random items, his postings on this list indicate
that he installs PBXes and resells VoIP services, and then his private
e-mails
Hi all,
I'm sure this is a stupid question, but is there a way to check your
voicemail by calling your extension from the outside? When I call my
own extension from outside and hit pound or star, it just stops my
greeting and gives me the beep. I'd like to call my extension and
press a key
No update on unicall and 1.4?
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Anton Krall
|Sent: Tuesday, December 26, 2006 6:15 AM
|To: asterisk-users@lists.digium.com
|Subject: [asterisk-users] 1.4 and unicall
|
|Guys, anybody knows
Thiru,
You can connect them by SIP or IAX, it depends on what kind of media you
will need to transfer. For audio only IAX is ok, if you 're going to use
video, SIP is the option.
Carlos Alperin
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thirumal
Saminathan
Phil,
Add this to your extensions (I have mine in a macro)
exten = a,1,VoicemailMain(${ARG1}); If they press *, send to
Voicemail
so it should look like...
exten = s,1,Dial(${ARG2},13,${ARG3})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten =
On Thursday 28 December 2006 13:33, Matt wrote:
Hi,
Well, in our case, it seems that the issue was being caused by
announcements. That is, someone in QUEUE1 would be waiting 15
minutes.. and QUEUE2 would be waiting 5 minutes.
Yep we noticed this too - it's a rather unfortunate side-effect;
I asked the same a while ago, without any kind of conclusive answer.
But you have to consider that these are special dates
I just spent all night studying/modifying mfcr2.c to my needs but
I've never looked at the unicall code or the asterisk channel API.
With respect to MFC/R2, and
So no one else is having issues with MySQL and 1.4? I'm the only one?
-Original Message-
From: Savoy, Kevin - Williston, ND
Sent: Wednesday, December 27, 2006 2:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] cdr_addon_mysql.so did not
I am using the Background() function to ask for the extension, but the
extensions are in a different context. Is there a way to tell Background()
to look for the entered extensions in another context other than the
currently running one?
Thanks.
Keith
I am using the Background() function to ask for the extension, but the
extensions are in a different context. Is there a way to tell Background()
to look for the entered extensions in another context other than the
currently running one?
in that context you can do
include = other-context
hth
If you type show application background on the *CLI, you can see all
the options listed there. The last optional argument is the context
that you want to use to look for extensions.
On 12/28/06, Time Bandit [EMAIL PROTECTED] wrote:
I am using the Background() function to ask for the
Ok so I'm the only one not getting this to work. Maybe I'm doing
something wrong. Here is the installation I'm using. Install Fedora Core
4 and do all the updates through yum. Then I install zdlib-devel,
openssl-devel, newt-devel, gcc, gcc-c++ and then mysql and
perl-DBD-MySQL all using yum
I hope so, he is the only guy working on mfcr2 right now.
I have unicall working on 1.2 perfectly but if there will be no unicall
support for 1.4, that would be a show stopper unless we use a mfcr2
converter... anybody knows any? Something that can convert mfcr2 to pri?
|-Original
Savoy, Kevin - Williston, ND wrote:
Ok so I'm the only one not getting this to work. Maybe I'm doing
something wrong. Here is the installation I'm using. Install Fedora Core
4 and do all the updates through yum. Then I install zdlib-devel,
openssl-devel, newt-devel, gcc, gcc-c++ and then mysql
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Savoy, Kevin - Williston, ND
Sent: Thursday, December 28, 2006 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: FW: [asterisk-users] cdr_addon_mysql.so did not
On 2006-12-24 00:35:06 -0800, Martin Joseph [EMAIL PROTECTED] said:
I have a spiffy new gateway which seems quite promising.
It's the Audiocodes MP114 FXS_FXO (2 of each).
I have got it configured and working reasonably well, but have a couple
of issues.
1) Asterisk 1.2.13 voicemail seems
Ok so something is missing. I get the below for those two lines.
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... no
I even installed the mysql-devel as Bradley Watkins suggested and still
it says no. What do I need to make that say yes?
Thanks
Each time I tried to update the firmware on two HT-496 boxes I got
Timeout error sending .bin from (192.168.1.94), 0 bytes
Then,
Transmit error while sending to 192.168.1.94. The connection is reset by the
remote side.
I tried on my LAN, and at last with a crossover cable between my
Can someone tell me how Asterisk handles music-on-hold between servers?
Documentation for this is non-existent.
Lets say user A, who is registered on pbx1, calls user B, who is registered on
pbx2.
1. User A puts user B on hold. The moh that is played to user B should be
specified according to
Tzafrir wrote:
On Thu, Dec 28, 2006 at 12:35:57PM +1300, Ray Jackson wrote:
Dan,
I have IMAP support working now with Courier IMAP. Since Courier
(and
probably Dovecot) do not support a single authuser connection that
may
access any mailbox, you have to omit the 'authuser' and
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I
Hi list!
I'm totally fed up with bristuff (or it's instability with a simple HFC-S
card), 2 out of 3 times when people try to call they get the information
tone that the number is not connected.
I would like to try vzaphfc and I am looking for information on it.
From previous posts I found
Rob,
Interestingly enough, I'm using that same sample macro, and that line is
indeed in there, yet when I hit *, I hear the tone to leave a message.
Any ideas?
Phil
Phil,
Add this to your extensions (I have mine in a macro)
exten = a,1,VoicemailMain(${ARG1})
Hello Everybody,
Since I upgraded to 1.4 I always get the difficulties as below, which I have
never had in 1.2:
[Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by
202.153.128.34 (format g729)
[Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729
[Dec 28
Hi List,
Hope everyone is recovering from the festive season :) (ok we still have
new years i guess!)
Anyways, I was wondering if anyone has had any successful dealings with
WiFi phones and operation with '*' at all?
I've been keeping my eye on the LinkSys WIP330 (
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI rtp debug
RTP Debugging Enabled
-- Executing Dial(SIP/xlite-007918f0, SIP/snom) in new stack
-- Called snom
-- SIP/snom-00797110 is
I bought a WIP300 to test and it was aweful.
It would either not register a keypress or register it twice.
It would also freeze up few minutes at a time.
It looks like the WIP330 has a new keypad, so maybe that problem is gone.
The WIP300 worked with asterisk, but I can not recall the quality at
Jay,
I had a similar issue recently... My filename had more than one .
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no .
- Upgrade sox to the latest version which fixes the funny behaviour
I recommend the hitachi wifi phones for use with asterisk.
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: (678) 248-2637 X: 1500
Direct: (678) 229-1809
http://www.sheltonjohns.com
**Sent from my mobile phone**
-Original Message-
From: Steven [EMAIL PROTECTED]
To:
Try setting in sip.conf:
nat=route
This tells asterisk to send all responses back to where the inquiry came
from rather then from the info contained in the sip packet.
Good luck,
Mark Coccimiglio
IS Director
Payroll Services Hawaii, Inc.
http://www.psh-inc.com
Elpidio Ramos wrote:
This
Found problem
xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't
know how to change this at xlite
venus*CLI
-- SIP read from 192.168.100.20:60726:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK02d4cc64;rport=5060
Contact: sip:[EMAIL
On 21:04, Thu 28 Dec 06, Remco Barendse wrote:
Hi list!
I'm totally fed up with bristuff (or it's instability with a simple HFC-S
card), 2 out of 3 times when people try to call they get the information
tone that the number is not connected.
I would like to try vzaphfc and I am looking
On 12/28/06, Bryan M. Johns [EMAIL PROTECTED] wrote:
I recommend the hitachi wifi phones for use with asterisk.
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: (678) 248-2637 X: 1500
Direct: (678) 229-1809
http://www.sheltonjohns.com
**Sent from my mobile phone**
-Original
Hi Wayne,
I was a very lucky guy this christmas, and received a D-Link DPH-540.
Despite the very first gen feel of the phone, I have been very impressed
so far.
You are correct in thinking that it can act as an extension external to your
network. So long as the place you're in has a decent
I agree, he sent me one off list, too - making all kinds of allegations of
my sexual preferences. I sent him a link to AA, DrPhil, National Institute
of Mental Health and suggested he get some help.
On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote:
Of course everyone is allowed to use VoIP...
Hi,
I have switched a while back from chan_capi to chan_misdn. When the
number is dialed and the phone is then picked up everything works just
fine. Some users however FIRST pick up the phone and then start to
dial... I did not get this to work with misdn.
When two digits have been dialed,
Well I'll be. That fixed it nicely. I was adding the .gsm extension
myself not realizing that Asterisk did it as well. Removing my addition
fixed the problem.
Thanks a ton!
Jay
Ex Vitorino wrote:
Jay,
I had a similar issue recently... My filename had more than one .
(dot / period)
-Ursprüngliche Nachricht-
Von: Wayne [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 28. Dezember 2006 22:20
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
Hi List,
Hope everyone is recovering from the festive season :) (ok we
On Thu, Dec 28, 2006 at 09:04:40PM +0100, Remco Barendse wrote:
Hi list!
I'm totally fed up with bristuff (or it's instability with a simple HFC-S
card), 2 out of 3 times when people try to call they get the information
tone that the number is not connected.
I would like to try vzaphfc
Hi,
We just upgraded to 1.4 and I'm noticing weird issues. I have noticed
that asterisk stops running and I need to restart in order for us to
receive calls. We receive our calls via a local sip provider over a
dedicated T-1. We never have had an issue before until the upgrade to
1.4. It
On Thu, 28 Dec 2006, Wayne wrote:
Hi List,
Hope everyone is recovering from the festive season :) (ok we still have new
years i guess!)
Anyways, I was wondering if anyone has had any successful dealings with WiFi
phones and operation with '*' at all?
I've been using an UT Starcom F1000G
On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote:
vzaphfc is not a complete replacement of bristuff. It replies on most of
it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI
driver for HFC-s-based PCI cards.
Further, if you're looking for 'something else' re:
Folks,
I have been trying to install Zaptel 1.4.0 on my SuSE 10.0 box with kernel
2.6.13-15.12. I have installed the kernel sources and run make
cloneconfig and make prepare. I have run ./configure but make linux26
is failing with the following error:
hawk:/tmp/zaptel-1.4.0 # make linux26
hello
this is the error
chan_vpb.cc: In function \u2018void mkbrd(vpb_model_t, int)\u2019:
chan_vpb.cc:1530: aviso: la dereferencia de punteros de tipo castigado
romper las reglas de alias estricto
chan_vpb.cc: In function \u2018ast_channel* vpb_new(vpb_pvt*,
ast_channel_state, char*)\u2019:
As if we needed more proof that Bochter was a screw-ball... He's now accused
me of being the owner of TRXTel. Not that we needed proof he wasn't actually
a PI, but in case anyone had any doubts, read the thread.
Alex
-- Forwarded message --
From: Al Bochter [EMAIL PROTECTED]
Here's what he sent me after I told him to shut the up. I kind of
wonder if he's just trying to generate traffic at certain sites and it's
going to generate ad revenue for him in some lame scheme. Oh well:
You could be using an older version of Asterisk that doesn't support it?
On 12/28/06, Phil Finkler [EMAIL PROTECTED] wrote:
Rob,
Interestingly enough, I'm using that same sample macro, and that line is
indeed in there, yet when I hit *, I hear the tone to leave a message. Any
ideas?
Hi all, as good?
I am trying to go up a board TE110P with link E1 ISDN PRI to establish
connection with a central office Siemens HiPath 4000. But I am having the
following errors:
Server1:~ # asterisk -r
Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer
I don't think if somebody making upgrades for the unicall in accordance to the
latest version of Asterisk. The latest patches of unicall and MFCR2 that I saw
is still for Asterisk ver. 1.2.0. Haven't see any patches for latest version
yet.
This what making me afraid of going to upgrade our
Can someone tell me how Asterisk handles music-on-hold between servers?
Documentation for this is non-existent.
Lets say user A, who is registered on pbx1, calls user B, who is registered on
pbx2.
1. User A puts user B on hold. The moh that is played to user B should be
specified according to
Or more likely the tone may not be getting to asterisk. What FXO are you
using? External FXO's like the SPA3000 often need to be set to 'inband'
DTMF - both in sip.config and in the device's config and be sure to
restart Asterisk after doing this..
Easiest way to test this is to call yourself
On Fri, Dec 15, 2006 at 06:32:19PM -0800, Yuan LIU wrote:
When booting Ubuntu 6.06.1 (Linux 2.6.15-27-386), wcfxo would load but not
configure. I have three ways to manually force wcfxo to configure: 1)
ztcfg, 2) modprobe -f wcfxo, or of course 3) unload and reload wcfxo.
Each works equally
I am after a long day of work, it felt realy good to laugh a bit.
On 12/28/06, Tom Lynn [EMAIL PROTECTED] wrote:
Here's what he sent me after I told him to shut the up. I kind of
wonder if he's just trying to generate traffic at certain sites and it's
going to generate ad revenue for him
On Thu, 28 Dec 2006, Gavin Hamill wrote:
On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote:
vzaphfc is not a complete replacement of bristuff. It replies on most of
it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI
driver for HFC-s-based PCI cards.
Further, if
On Thu, 28 Dec 2006, Michiel van Baak wrote:
When you found out stuff, specially how to make stuff with a
simple HFC-S card stable please let me know.
We are not deploying them cards anymore because we never get
it stable.
Real simple setups can be done with a FRITZ!PCI card, but I
really
Hi,
In my ip phone is voicemail indicator, and also is a voicemail button (to
access to voicemail server and ant to listen voicemail). My question is how
to configure this button. In configuration I need to enter URL. What is the
syntax of this URL, that IP Phone could fetch this voicemail from
On 12/29/06 06:04 Hans-Jürgen Brand said the following:
Found problem
xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't
know how to change this at xlite
have you tried nat=yes in sip.conf for the peer ?
--
Regards, /\_/\ All
Trouble is this (promising) phone is not distributed everywhere, at least,
not here in France, yet.
I couldn't get any reason from Siemens France.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Savoy, Kevin - Williston, ND wrote:
Ok so something is missing. I get the below for those two lines.
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... no
I even installed the mysql-devel as Bradley
63 matches
Mail list logo
