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
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
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
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
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
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.
___
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
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:
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
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
, 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
- Original Message -
From: Lal, Deepak (Contractor) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 24, 2003 3:09 PM
Subject: [Asterisk-Users] SS7 signaling/Softswitch
I'm confused a bit about the following and was hoping to get some answers
on
this group - What is
Hi all:
I've no response for the last question with the same subject. Please excuse
me for the extreme length of this mail, but I send 2 SIP traces.
I have problem with * and 5300, when the incoming and outgoing call are
routed thru the same SIP gateway (AS5300). Do I need to set an special
Someone has proven chan_skinny with Cisco 7910?
I've got some problems with dtmf relay:
Oct 23 14:58:30 WARNING[-1533859520]: File
chan_skinny.c, Line 1710 (skinny_indicate): Don't know how to indicate condition
14
Thanks in advance,
Gus
Rich:
Please see if festival_server is running as specified in:
http://www.marko.net/asterisk/archives/0209/0389.html
==
export PATH=$PATH:/usr/src/festival/bin
/usr/src/festival/bin/festival_server
==
Or test festival in bash...
Regards,
Gus
- Original
Do you have ztdummy or zaptel device in your system?
- Original Message -
From: Panny Malialis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 1:43 PM
Subject: [Asterisk-Users] Meetme
Is app_meetme broken?
I seem to get invalid conference number all the
Please see if this helps.
Regards,
Gus
noc2pbx*CLI show application MeetMe
noc2pbx*CLI
-= Info about application 'MeetMe' =-
[Synopsis]:
Simple MeetMe conference bridge
[Description]:
MeetMe(confno[|options]): Enters the user into a specified MeetMe
conference.
If the conference number
Mickey:
At least in some European countries and southamerican environments, you
can't send your own ANI.
For example: if you have a PRI with the number: 5288 to 52880099, you
could send as ANI any number between 000 and 099. The fixed part will be
transmitted by your PSTN switch to the
Hi all:
I've just purchase some licences of G.729 codecs,
and I like to bring up * using /etc/rc.d/init.d script.
Does anyone knows how to start in the "old"
way?
Thanks in advance,
Gus
I will do.
Thanks a lot.
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 16, 2003 6:28 PM
Subject: Re: [Asterisk-Users] Starting * with G729 licences
On Thursday 16 October 2003 16:09, CW_ASN - Gus wrote:
Hi all:
I've
PROTECTED]
Sent: Thursday, October 16, 2003 6:13 PM
Subject: Re: [Asterisk-Users] Starting * with G729 licences
check 'screen -d -m asterisk -vvvcng'
regards
Martin
On Thu, 16 Oct 2003, CW_ASN - Gus wrote:
Hi all:
I've just purchase some licences of G.729 codecs, and I like to bring
codec The Voiceage part of the codec breaks it.
regards
Martin
On Thu, 16 Oct 2003, CW_ASN - Gus wrote:
Martin:
This works ok. Doing a 'ps ax | grep aste' shows:
3071 ?S 0:00 SCREEN -d -m asterisk -vvvcng
3072 pts/2S 0:02 asterisk -vvvcng
This means
I don't know how US works in matter of PRI and his standards, nor C7 ANSI
level, but I think that don't have a lot of differences with european
standard.
Make sure that you are sending ANI in setup message, and presentation byte
is set to Allow. If you don't know, please send a 'pri debug'.
Also: Which codecs are you using?
- Original Message -
From: TeleSIP [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 16, 2003 7:07 PM
Subject: Re: [Asterisk-Users] MOH and VAD
Also I found that ATA use silence supresion on the second phone even if
they have
Yes, I'm using that... but I will
change:
- daemon /usr/sbin/asterisk
+ daemon screen -d -m asterisk -vvvcng
Thanks,
Gus
- Original Message -
From:
John Haigh
To: [EMAIL PROTECTED]
Sent: Thursday, October 16, 2003 6:55
PM
Subject: RE: [Asterisk-Users]
Alastair:
At least in some European countries and southamerican environments, you
can't send your own ANI.
For example: if you have a PRI with the number: 5288 to 52880099, you
could send as ANI any number between 000 and 099. The fixed part will be
transmitted by your PSTN switch to the
Marcel:
Generally, the ton for A subscriber is setted as National Number, and ton
for B subscriber is setted as local.
In the specific case for Siemens EWSD (wonderful switch), it does a
conversion a digit using an internal table, this tables do this:
EWSD Side
Walker:
sip show channel refers to a Call ID:
noc2pbx2*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
Format
172.16.254.620341522910 3607139911@ 00101/3 0ms ms ALAW
1 active SIP channel(s)
Then, you could see the details:
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