Title: Message
Please don't cross message between
lists.
- Original Message -
From:
Mayank Mishra
To: [EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 6:40
AM
Subject: [Asterisk-Users] How to get Call
Details Records
HI,
Can anyone please
tell
Don't cross messages between lists.
Anyway, be more specific.
- Original Message -
From:
shabanip
To: [EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 12:02
PM
Subject: [Asterisk-Users] Whoa I'm
owned but found ??
I get this message at CLI.
is there a well-written, easy to follow, voicemail setup guide for
asterisk?
No, but you don't need setup guide. See wiki.
Regards,
Gus
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Trace from their analyzer attached.
Can they send an EWSD trace???
switchtype was already set to euroisdn, so that shouldn't be the problem.
I first configured pridialplan=unknown, but the telecom partner asked me
to change the TON (type of number) to unknown, and the NPI to
Hi all,
We're trying to hook up our Asterisk config (Card: TE410P) with a
Siemens EWSD switch. The link is ok on both ends (green), with no errors.
The problem is when we try to make a call from our side (via call
files), we get the pri/E1 error
Ext: 1 Cause: Temporary failure (41),
hi!
I need to pass the CLI for my outgoing ISDN PRI call from * box.
here's the ISDN protocol debug.
Q.931
Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Presentation permitted, user
number passed
I believe that 'ast_data' is the solution to this problem, and will
probably obsolete mysql friends. However, I could be incorrect in that
manner. There are folks on this list who would be much better informed to
say whether or not it will obsolete mysql friends.
-Chris
I did not tests
Hi,
I've forgotten the command to add a vm box, and searching google and wiki
I'm
surpriced I cannot find it. I'd love to know where this is written, so I
can
see how I managed to miss it!
- --
Steve
Look for your controb/script directory. The script is called 'addmailbox'.
Regards,
the
chan_oh323. The asterisk now can start again. :)
And Gus, could you tell me what's the meaning of
IMHO? I can't find the topic about IMHO in WIFI.
Thanks a lot!
Best Regards
Rui
--- CW_ASN [EMAIL PROTECTED] wrote: Hi,
After I install openh323, the asterisk cann't
work
. I choose no load the chan_oh323. The asterisk now
can start again. :)
And Gus, could you tell me what's the meaning of IMHO? I can't find the
topic about IMHO in WIFI.
Thanks a lot!
Best Regards
Rui
--- CW_ASN [EMAIL PROTECTED] wrote: Hi,
After I install openh323, the asterisk
As I explained to you before we use it for our customer service in call
(B center and implemented in many call centres which really makes $.
(B
(BAll this stuff to do a simple call queue system??? Man, You need to read
(Bwiki. Read agents.conf and queue.conf before to begin a war here...
Check wiki for patch... maybe it's you best option.
Regards,
Gus
- Original Message -
From: Ethan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 13, 2004 4:22 PM
Subject: [Asterisk-Users] Rotary phones? (No, I'm serious)
Will the FXS cards that work with asterisk
Hi,
After I install openh323, the asterisk cann't work
anymore. Asterisk failed in loading chan_oh323. I
cann't deleted the openh323 package, so the only thing
I can do is to reinstall Asterisk. I checked out the
asterisk and make install Astersik without installed
openh323, but when I
Is there any way for me to add myself to a call queue from outside of my
Asterisk Box?
For example,
I have a queue set up on my asterisk box, and I want to call it on my Cell
Phone, then add myself to the queue and hang up.. When a call comes into
the
queue, I want it to be forwarded to my
It's included in CVS. I'm using it from there!
Anyway, the patch is 214. Look
http://bugs.digium.com/bug_view_page.php?bug_id=214
Regards,
Gus
At 00:35:41, CW_ASN wrote:
Please try CVS, AFAIK patch 214 doesn't included in stable branch.
But I need to apply some other patches too
Anyone have any experience using gr303?
May have a need to interface * to a Siemens Class-5 CO for pstn
trunking (inbound and outbound).
Rich
I assume Siemens Class5=EWSD.
EWSD is compatible with GR.303, and AFAIK it works with special national
project.
Which software version (APS) and
Man, just provide us more info... debugs, logs, anything.
You don't need to pay for help.
Regards,
Gus
- Original Message -
From: Stuart Baggs [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 6:57 PM
Subject: Re: [Asterisk-Users] prepaid application
Could
Please try CVS, AFAIK patch 214 doesn't included in stable branch.
Regards,
Gus
- Original Message -
From: Robin Calmegård Siurua [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 7:10 PM
Subject: [Asterisk-Users] Patch for call queues?
I'm looking for the
You have problems with pgsql. Check it.
Regards,
Gus
- Original Message -
From: Hekuran Doli [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 5:27 PM
Subject: [Asterisk-Users] Midifyed-Prepaid-Application
Hello.
I have compile asterisk with modifyed prepaid
Send traces.
- Original Message -
From: Aimable [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 6:28 AM
Subject: [Asterisk-Users] Problems with PRI with T410 messages
Hi all,
I have a box running asterisk with T410 connected to a Nortel DMS 100
switch
and
This is a problem I pointed out to Digium a while back, but I am not sure
Markster understood the issue and I didn't really have the time to follow it
up. It does need fixing though, as it is a major drawback in the current
architecture.
Rgds
Tim
Hi all,
I have a box running asterisk with
I do not believe you are correct. We see CALL PROCEEDING in both
directions as part of the normal ISDN call setup process. See trace
below.
Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL
PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below:
Or compile the .so with -lpq option.
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 5:06
AM
Subject: Re: [Asterisk-Users] Prepaid
application error
Hi, you have to launch the
script prepaid-make.sh in the
Obviously, you have seen very few OM interfaces.
Regards,
Gus
- Original Message -
From: W. Kevin Hunt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 09, 2004 6:26 PM
Subject: RE: [Asterisk-Users] Re: NetworkWorld article on Open Source
Telephony
I happen to feel that
- Original Message -
From: Fabio Donaggio
To: [EMAIL PROTECTED]
Sent: Friday, May 28, 2004 6:16 AM
Subject: [Asterisk-Users] Asterisk addons
Hi to all!!
Is there another method to download asterisk addons???
Thanks
F
Man! Try to investigate for yourself! Use google!
- Original Message -
From: Fabio Donaggio
To: [EMAIL PROTECTED]
Sent: Friday, May 28, 2004 12:52 PM
Subject: [Asterisk-Users] Fw: Asterisk and MySQL
Hi!
It's all ok with CVS login...I download asterisk-addons.
I would try to store sip friends in MySQL database and also the
Paste your extensions.conf
Check the answer command if you're running IVR of special services.
- Original Message -
From: Jorge Verastegui
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 8:46 PM
Subject: Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA
When i make a call from
make webvmail
from your source directory. Then, point your browser to:
http://your_ip/cgi-bin/vmail.cgi
Regards,
Gus
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Kurt
Enviado el: Miercoles, 21 de Abril de 2004 12:36 p.m.
Para: [EMAIL PROTECTED]
Asunto:
No, you don't need to change permissions. Check in your voicemail.conf the
user password for accounts.
I don't know how vmail.cgi works with multiple contexts, or if you have
mysql/pgsql support with app_voicemail.
See http://www.voip-info.org/wiki-Asterisk+gui+vmail.cgi for more details.
http://www.voip-info.org/tiki-print.php?page=Asterisk+PBX+functions
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]En nombre de Kyle
HaganEnviado el: Martes, 20 de Abril de 2004 02:23
p.m.Para: [EMAIL PROTECTED]Asunto:
[Asterisk-Users] Extention
In fact, with EWSD V13 you can't remove CRC4 in PRI mode.
Regards,
Gus
- Original Message -
From: Storer, Darren [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 8:32 PM
Subject: RE: [Asterisk-Users] Siemens EWSD 13
Hi,
I had exactly the same symptoms today
When you see this message, try to kill mpg123 from another terminal (to stop
'Ouch...') and review the previous errors.
Regards,
Gus
- Original Message -
From: Simon Brown [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 28, 2004 10:37 PM
Subject: [Asterisk-Users] Broken
Try adding 'insecure=yes' in sip.conf.
Regards,
Gus
- Original Message -
From: Joao Carlos Moura [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 20, 2004 12:02 PM
Subject: [Asterisk-Users] Basic authentication
How can I settup a way for Asterisk doesn´t make any use of
You can't expect much help without data...
Post the last compile messages, platform, SO.
Regards,
Gus
- Original Message -
From: Hubert Kiyimba [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 15, 2004 5:31 AM
Subject: [Asterisk-Users] error, installing asterisk
I got
Alex:
In 'call' table stores call details.
'card' stores user pin (10 digits in original version)
'country' associates a short description with a long description of
destination.
'countryprefix' associates prefix (i.e. 1305) with short description (of
'country' table) and type of destination
See monastery, maybe help you (http://pbx.unslept.com/newstatus.php)
Regards,
Gus
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 08, 2004 6:27 PM
Subject: [Asterisk-Users] SIP - Receptionist
Hi All!
I am thinking about fork-lift-upgrading a
So put your hands on it and help to product grow.
Regards,
Gus
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 08, 2004 8:19 PM
Subject: Re: [Asterisk-Users] SIP - Receptionist
Monastery is neat as a monitoring tool. The console's we're
You must change the setwhentohangup function, see channel.c for that.
Someone wrote a patch to do this (see http://bugs.digium.com/).
Regards,
Gus
- Original Message -
From: Hans-Henrik Andresen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 07, 2004 12:31 PM
Subject:
This is wrongs. It's me who wrote the patch, it's available in CVS
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
Nopez i'm not
In that case, exists another patch from a guy called Klaus. I'm using this
patch since Dec2003.
Maybe helps, I don't know, but this is other alternative.
Its merged
Hi all:
I'm doing some tests with sip equipments, and
sometimes I see:
DEBUG[1150495040]: File chan_sip.c, Line 5077
(handle_request): Hm No sdp for the moemnt
Does anyone knows anything about this?
Thanks in advance,
Gus
Why people don't have al least some respect about regulations?
Sure that pridial=unknown solved that problem, but sadly you're overwriting
the main class of service indication in ISDN...
Unknown let to Class 5 switch manage (as the operator wish) understand
your messages.
The common sense shows
3.0.0 have some problems. Sometimes, ata answers to invite with Not found
or Busy here. This is a strange behavior.
I'm using now 2.16.2
Regards,
Gus
- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 12:56 AM
Subject:
Could you share your 3.0.0 config?
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 2:10 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
Hi,
Citeren CW_ASN [EMAIL PROTECTED]:
3.0.0 have some
://www.nxs.net/cisco_ata_186.htm
- Original Message -
From: CW_ASN [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 12:40 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
Could you share your 3.0.0 config?
- Original Message -
From: Florian
It must be:
exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]
Hope this helps,
Gus
- Original Message -
From: Anthony Law [EMAIL PROTECTED]
To: Mailing List Asterisk [EMAIL PROTECTED]
Sent: Friday, February 06, 2004 11:56 AM
Yes, lot of people use ztdummy.
- Original Message -
From:
Paul
To: [EMAIL PROTECTED]
Sent: Monday, February 02, 2004 12:49
AM
Subject: [Asterisk-Users] Meetme without
zaptel hardware
Has anyone had any success using
the ztdummy module and doing
How? Is written in CDR?
Regards,
Gus
- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 30 January 2004
:48 AM
Subject: Re: [Asterisk-Users] HANGUPCAUSE
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
HANGUPCAUSE is working fine here (cvs).
How? Is written in CDR?
CDRs contain BUSY when busy and NO ANSWER on the rest.
extensions.conf:
[provider
, CW_ASN - Gus wrote:
Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .
You could use my approach and combine it with the CDR userfield. Personally
I
would like a PRI_CAUSE variable to be set
Try with:
http://bugs.digium.com/bug_view_page.php?bug_id=214
Regards,
Gus
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 11:01 AM
Subject: [Asterisk-Users] app_queue and dialplan
Hello,
I`m trying to achive
Try with:
http://bugs.digium.com/bug_view_page.php?bug_id=214
Regards,
Gus
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 9:59 AM
Subject: [Asterisk-Users] app_queue and dialplan
Hello,
I`m trying to achive
If you don't have the licences for this codec, you can't playback files from
*.
If I'm not mistaken, * can be used to do codec passthrough between two
endpoints, but you can't use any application to interact with *, like
voicemail, directory, background or playback.
Regards,
Gus
- Original
The incoming call request Unrestricted and 64K, and this looks like ok, but
in the SETUP_ACK the called number parameters shows: Ext: 1 Progress
Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN
equipment.
In the most of cases, Information transfer rate = to '64 kbit/s',
RR--|
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
Gus
Gesendet: Donnerstag, 22. Januar 2004 17:24
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
The incoming call request
]
#12 RR--|
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
Gus
Gesendet: Donnerstag, 22. Januar 2004 17:24
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Please send your zaptel.conf to see what's going on.
- Original Message -
From: Daniel Bichara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 4:38 PM
Subject: [Asterisk-Users] ETSI PRI ISDN Signalling
Hi All,
I've bought a R2Adapter to convert R2Digital
Maybe Telefonica (the same from .ar) is not big enough!
By the sight Telefónica in Brazil is not very serious, in Argentina offers
ISDN in all country, for all kinds of teleservices... I'm sure of that.
___
Asterisk-Users mailing list
[EMAIL
CW_ASN - Gus wrote:
Ok, it's old and clunky, but in some countries like Brazil, Argentina and
China is the only alternative.
Only alternative??? Why is the only alternative? All mayor carriers in
Argentina and Brasil have PRI signalling, at the same price.
Sorry, but you
Ok, it's old and clunky, but in some countries like Brazil, Argentina and
China is the only alternative.
Only alternative??? Why is the only alternative? All mayor carriers in
Argentina and Brasil have PRI signalling, at the same price.
___
yes but PRI is not a trunk,
Not in all switches...
You have a Siemens EWSD (I know your company), if you change to V15 you can
treat the PRI like a route (and a lot of things more).
I have Siemens EWSD and Lucent 5ESS, and for 5ESS, the PRI is a route.
I see only one reason to use R2... only
See http://www.rad.com/ , TDM-over-IP solutions.
- Original Message -
From: Alexandru Coseru [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 19, 2004 6:56 AM
Subject: Re: [Asterisk-Users] SS7 over Asterisk ?
Maybe , I never tried TDMoE ...
Where can I found a
run * in console mode and send the
log.
asterisk -cv
- Original Message -
From:
Paul
To: [EMAIL PROTECTED]
Sent: Sunday, January 18, 2004 11:31
AM
Subject: [Asterisk-Users] No startup
after mpg123 install
After installing mpg123 * will no
Try with:
make webvmail
from source directory.
- Original Message -
From:
tony banks
To: [EMAIL PROTECTED]
Sent: Monday, January 12, 2004 1:45
PM
Subject: [Asterisk-Users] Issue -
vmail.cgi on Redhat 9 (Apache) ?
HelloI found related question on
Jess:
Try with:
Dial(SIP/[EMAIL PROTECTED],20,t)
Remove 'r' option from your dial command, maybe 'show application Dial' from
CLI could help you more.
Regards,
Gus
- Original Message -
From: Jess Magnaye
To: [EMAIL PROTECTED]
Sent: Friday, January 09, 2004 7:55 PM
Subject:
Sure, declare the queue and its timeout, then declare the same extension
with voicemail with n+1 priority.
exten = 2056,1,Answer
exten = 2056,2,Wait,1
exten = 2056,3,Queue(noc|t|||30)
exten = 2056,4,VoiceMail(u2056)
Hope this helps,
Gus
-= Info about application 'Queue' =-
[Synopsis]:
Queue
snip
Yes - the Wiki link above about call queues has the info and links that
you need to look at.
Also, could be great is you install a new patch, to add some great
functionalities to your call center. This path is located:
http://bugs.digium.com/bug_view_page.php?bug_id=214
Regards,
Gus
Sometimes Cisco nat changes the port, and * can't contact to ATA. I see that
behavior some days ago, and I can't resolve that. :(
Regards,
Gus
- Original Message -
From: Osvaldo Mundim Junior [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 9:15 PM
Subject: Re:
Are you using 1605 to do nat?
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 7:12 PM
Subject: Re: [Asterisk-Users] ATA call
I have ZERO problems with Cisco's NAT for SIP.
On Tue, 2004-01-06 at 13:42, CW_ASN - Gus wrote
snip
this is called Message Waiting Indicator (MWI) in asterisk. I
haven't set it up myself,
but from what I've seen there are a few parts:
1) setting a mailbox=1234 etc. in the extension definition in the channel
file
2) setting up the phone
Have a look around the wiki
And why you have two different entries for the same object?
Posting two times the same questions with other data will not help to
resolve the issue more quickly...
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Saturday,
- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 1:34 AM
Subject: Re: [Asterisk-Users] Call recording/SIP not loggin IN
My sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
disallow=all
If you are a person who likes all things easy, and if you don't need to know
nothing to be better professional, well, run now, and let us continue our
journey. Who cares? People likes you don't help to our community.
Regards,
Gus
- Original Message -
From: Me [EMAIL PROTECTED]
To:
Dear newbies,
As a newcomer to woodworking, you will not be welcomed with open arms.
First, you will find no documentation on how to make your completely
custom
ceiling-height cabinets perfectly the first time that your wife will
appreciate. Second, if you ask any woodworker for
It works for me with sip 2.15, 2.16.x and 3 versions.
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 29, 2003 6:42 AM
Subject: [Asterisk-Users] call pickup via *8 from ata186 (SIP)
Hello,
Does call pickup works with ATA-186
Shad:
Using the AddQueueMember. Launching this command 3
times in different queues, logs one phone to that 3 queues...
*CLI show application AddQueueMember
-= Info about application 'AddQueueMember' =-
[Synopsis]:Dynamically adds queue members
[Description]:
Easier but poorly documented solution. AgentCallbackLogin()
AgentCallbackLogin delivers callo for a logged in agent to an extension.
- they continue to get calls until they log out (by logging in to a null
extension (pressing # when prompted for extension)
But AgentCallbackLogin remains the
McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 26, 2003 6:10 PM
Subject: Re: [Asterisk-Users] prepaid app
CW_ASN wrote:
Send an email to Bartosz, he has app_prepaid. You will need to work a lot
with C (i'm learning) and pgsql, but is very nice app.
First off he
Doesn't matter. If he uses the C API he he bound by the GPL or he has
to pay digium's fees for non-gpl.
Who in the hell said that is not GPL? I'm not sure about the licence of this
app, but in the .c code shows a nice GPL...
Maybe this 2 lines makes your life easier...
* This program is
How about to build an ip phone with this IC?
http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId=
969path=templatedata/cm/general/data/bband_ipphone_tnetv1001
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December
Skinny phone functionality is 'richer' than SIP phone functionality.
First
off, on a skinny phone, in hands free mode, you can start dialling and the
phone will automatically go off hook. Sip requires you to manually hit
the
speaker button, hit new call, or pickup the phone before dialling.
Which events do you refer?
Regards,
Gus
- Original Message -
From: Jonathan Tew [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 12:25 PM
Subject: Re: [Asterisk-Users] Asterisk + CRM
We're starting to integrate * with our customer service software.
Guys, I'm using RH9 with vmail.cgi without any modifications... I'm just do
a 'make webvmail' after 'make install'... I don't have any troubles...
Regards,
Gus
- Original Message -
From: Carlton J. O'Riley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 4:23 PM
Try something like this:
exten = 2060,1,Answer
exten = 2060,2,Wait,1
exten = 2060,3,Monitor,wav|algo
exten = 2060,4,Meetme,1|ps
Regards,
Gus
- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 8:58 AM
Subject:
algo is a file where app write a wav data. In spanish, algo means
something... :)
Gus
-= Info about application 'Monitor' =-
[Synopsis]:
Monitor a channel
[Description]:
Monitor
Used to start monitoring a channel. The channel's input and output
voice packets are logged to files until the
Try with another codec different than G.723. Use GSM o G.711 for this.
You could disable G.723 in your sip.conf
disallow=all
allow=gsm
allow=alaw
allow=ulaw
Hope this helps,
Gus
- Original Message -
From: Hachy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003
Title: Mensaje
Fijate en los 'voice codecs' de los
dial-peers.
- Original Message -
From:
Sebastian Nocetti
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 12:41
PM
Subject: [Asterisk-Users] Media
Negotiation Failed
Hi, I have this
scenario
Did you record the messages as gsm format?
- Original Message -
From: Larry D. Black [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 6:33 PM
Subject: [Asterisk-Users] menu prompts and voice mail greetings.
What program do you use to record menu prompts and
Yes, is posible.
- Original Message -
From:
marin
blu
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 3:22
AM
Subject: [Asterisk-Users] Manager
Server
Hi,
Is it possible to control * fromthe TCP Manager Server in order to
support CRM
Just replace Voicemail by VoiceMail2 and that's all.
Note that new voicemail.conf is a bit different than old voicemail.conf.
Regards,
Gus
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 4:44 AM
You could use DISA app.
exten = 2101,1,DISA,/opt/pass.txt|default
Where:
/opt/pass.txt is a plain text file with password list.
default is a destination context.
Anyway, please do 'show application disa' from CLI.
Hope this helps,
Gus
- Original Message -
From: Jacky Chen [EMAIL
Here is my example. I'm using a lot of times a day.
?php
$socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: blabla\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket,
, 2003 5:21 PM
Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch
CW_ASN - Gus wrote:
Anyway, in certanly implemetations you don't need CCS7 to connect to CO.
You
always can connect with PRI... same speed and same functionalities to
user
side. In fact, CCS7 is the support for ISDN-PRI
I didn't know it... excellent!
- Original Message -
From: Thorsten Lockert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 6:36 PM
Subject: RE: [Asterisk-Users] Music Onhold Configuration
MPG123 is not included in Asterisk...
Download the package:
Close. Normally, at least in Qwest-land, third-party VM provider systems
dial
into the switch and give it a DN and a MWI on-or-off command. If the DN
is
serviced by that switch, it turns the message waiting indicator (stutter
dialtone, MW light or both) on or off. If the number is on a
Yes, its true. Contact to [EMAIL PROTECTED]
- Original Message -
From: tad [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:41 PM
Subject: [Asterisk-Users] dialogic support
i am new to asterisk, and looking to develop an application using a
dialogic card. as
What kind of gateway are you using? Did you set dtmf-relay in that gateway?
Regards,
Gus
- Original Message -
From: Steve Dolloff [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:50 PM
Subject: [Asterisk-Users] passing digits for voicemail from sip gateway
I
Lars:
Anything you want is possible to do with Asterisk... the matter is how much
time you want to spend to build that applications... I think that is posible
to do that with AGI scripts...
Regards,
Gus
- Original Message -
From: Lars Fredriksson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
MPG123 is not included in Asterisk...
Download the package:
http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
Install using:
rpm -ivh mpg123-0.59q-1.i386.rpm
Copy the file mpg123 from /usr/local/bin to /usr/bin
That's all...
Please read the posts, this issue was treated
Sometimes, if * dies, mpg123 keeps running and eats all memory.
Try to stop *, kill all mpg123 instances and try again.
Also, you can modify your start script to kill all mpg123 instances before *
starts 'killall -9 mpg123'
Regards,
Gus
- Original Message -
From: TeleSIP [EMAIL
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