Re: [Asterisk-Users] Downgrading Asterisk
On Tue, 25 May 2004, jo wrote: Sorry, no solution but same problem. Downgrading brings this message on Suse9.0, 2.4.21: [app_txtcidname.so]May 25 23:28:42 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/app_txtcidname.so: undefined symbol: ast_get_txt May 25 23:28:42 WARNING[16384]: loader.c:408 load_modules: Loading module app_txtcidname.so failed! app_txtcidname.so is left over from your test of the new version. Delete it. Better - delete everything in /usr/lib/asterisk/modules and re-make install the version of Asterisk you want to use. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FYI: Cisco firmware 7.1 released
Cisco has version 7.1 of their SIP firmware for the 79x0 phones. They advertise no new software features, but it does include bugfixes for a number of things. I know there was a discussion about the 0.4sec delay, which is said to be resolved in this firmware (CSCed48311: Media takes 0.4 sec to be set up) I'll add to the above... Looks like v7.0 fixed 26 known bugs (not all of which pertain to the sip version), and v7.1 lists 28 resolved caveats (some of which appear to be duplicate descriptions of those noted as fixed in v7.0). V7.1 lists the following as open caveats: - SIP: 79x0 phones are not escaping reserved characters in URI/URLs - SIP: Need to CACHE cycle thru multiple DNS entries for FQDN Type A - SIP: Phone are not sending ICMP port unreachable - SIP: Inconsistent behaviour of AutoComplete feature on 79x0 phones - Anonymous Call Rejection returns wrong Response code - SIP: 79x0 ignores dst_start_time parameter when configured - SIP: Tx INVITEs to wrong port number after Rx 302 Moved Temporarily - SIP: CallerID Blocking needs to change more values to ANONYMOUS Initial tests with v7.1 and CVS-HEAD-05/26/04 look good thus far. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 11 instead of Star
Hi, -Original Message- OK. Well, here are a couple of newbie-type thoughts on the whole Vertical Service Code (CLASS) hard-codings. *snip* Yes, having a way to redefine class codes would be excellent. Especially since 'industry standard' only means 'industry standard in greater USA area'. Europe generally has totally different codes. Or has this already been discussed to death? It has. Mark has indicated the proper way to deal with this is to create a generic codebase to do it. At the moment these codes are defined in each individual channel. Not a very maintainable option. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Tony - This sounds great. Are you monitoring the line constantly for the inbound caller ID or are you somehow detecting the polarity reversal? Look forward to trying out your solution! I think you will have a lot of happy bunnies here in Blighty, as the current lack of caller ID on our BT lines is the thing that makes Asterisk less viable for most home users If it works and is stable, will you disclaim your code so that it will get merged into the main CVS? There should probably be a couple of settings in zapata.conf for the caller id coding scheme to be used for each card calleridtones = (V23, Bell or DTMF) calleridarrives = (afterring, afterrev, etc) since a lot of people here in UK have a line from BT and a cable co line, where the cable co either uses Bellcore after 1st ring, or V23 after 1st ring. So you need to be able to chose the method for each line. What a mess, eh? Rgds Tim Tony Hoyle wrote: Anyway, the caller ID patch is finished and works really well - it was easier than I thought... once you've decoded the V23 data the packet format is the same as the US Bell system. I've got some more testing to do to make sure it doesn't cause asterisk (or the kernel) to explode then I'll stick it on my website for general consumption/laughter/criticism Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NMI occures while loading the zaptel module
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, We are on a point, where we have no further soft options to resolve the following problem: While loading the zaptel and wct4xxp module, the following NMI occures: Uhhuh. NMI received. Dazed and ... You probably have a hardware problem with your RAM chips. The NMI is not a real problem, asterisk just starts fine and everything is ok so far. But we are running diagnostic software from HP to detect hardware problems. And this software produces a kernel panic because of this NMI from the zaptel module. We can observe this NMI on two machines of the same type and software configuration. Hardware: HP DL380 G3, 1024 MB RAM, 2x Xeon 3.2 Ghz /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 81676 0 0 0IO-APIC-edge timer 1: 19 0 0 0IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 8: 1 0 0 0IO-APIC-edge rtc 24: 696818 0 0 0 IO-APIC-level t4xxp 29:495 0 0 0 IO-APIC-level eth0 30: 60529 0 0 0 IO-APIC-level cciss0 NMI: 1 0 0 0 LOC: 81544 81543 81543 81543 ERR: 0 MIS: 0 Software: RedHat Linux 7.3 Professional, the whole bunch of HP Software What we have done so far: - - tried kernel version 2.4.18-3, 2.4.25, 2.4.26 - - tried zaptel stable and latest - - disabling HyperThreading in BIOS - - removed the second processor - - run memtest86 for about 7 hours - no problems - - disabling SMP support in the kernel - - enabled/disabled APIC-Support - - contacted support at digium, they say it is a problem in the kernel, not a problem with their module I have no idea except, change the operating system or the hardware or fix the software bug - if it is a software bug. Any hints, advices? - -- Regards, Cyrill Rüttimann - -- innovate IT AG Tel. 01 430 54 50 Badenerstrasse 808 Fax. 01 430 54 51 8048 Zuerich Web: www.innovate-it.ch Switzerland EMail: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAtEwGYTILjDeCowIRAjS4AKCfsUt8LfhI1iE2KY8yoEE02d1KLwCght/Y 1lM8knB91GtWWPtzg1utUwo= =ZyA4 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Asunto: Re: [Asterisk-Users] Troubles with Kphone]
Well .. I'm now using Kphone 3.11 and alsa and everithing looks good.. but when i dial an extension i only hear and horrible ticking sound ... like a burned dial up modem ... i can see how the call initiates, and finishes in the console .. thanks for all Ivan -- Mensaje original -- From: Murali Krishnan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Troubles with Kphone] Reply-To: [EMAIL PROTECTED] Date: Tue, 25 May 2004 16:14:11 +0530 Original Message Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Organization: bk SYSTEMS (P) LTD., To: [EMAIL PROTECTED] References: [EMAIL PROTECTED] enano wrote: Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport CSeq: 6121 ACK To: sip:[EMAIL PROTECTED];tag=as12aab0bf From: ivan2 sip:[EMAIL PROTECTED];tag=7F6911ED Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0.2 Contact: ivan2 sip:[EMAIL PROTECTED];transport=udp res_search: NO result ! res_search: NO result ! SipClient: Sending to '192.168.0.3:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 192.168.0.3:19696 UDPMessageSocket::SetTOS: Operation not permitted CallAudio: OSS device already open (readwrite) anyone can help me ?? thanks Ivan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Check the following things. 1. Make sure your sound card is configured properly for record/playback - if not, do it with either kmix and test with gnome-sound-recorder 2. Make sure your identity is configured in sip.conf and extension.conf correctly 3. Make sure kphone is registered with Asterisk File-Identity - see whether 'Unregister' is there, (means you are registered ) 4. Watch for Asterisk Messages for any clue. ( asterisk -vc ) 5. Make sure the destination extension you are dialing from kphone has proper dialplan sequence in extension.conf 6. If you have OSS sound configuration, immediately switch to ALSA. - visit alsa-project.org and search docs for your card type. Compile and install the packages. ( this OSS would be the major headache if you are not getting sound ). If you are registered with Asterisk and your sound card is proper, and you configured your destination extension routing properly in extension.conf everything should work fine. Get back with success. Regards Murali Krishnan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phones
Hi everyone, I have to test few models of SIP IP phones with Asterisk. I have seen on voip-info.org that there are lot of phones that work ok with Asterisk. But, I want to ask for suggestion - which models are the best for Asterisk? I would appreciate if I can choose 4-5 models for test. Tomica Crnek
RE: [Asterisk-Users] sip phone problem
--- Antonio Diego [EMAIL PROTECTED]
escribió: Hi,
First you need to upgrade to the latest CVS and
then
insert a second /
third priority line with hangup in the dialplan.
Regds
Vivian Alan
Hi Alan, thanks for your help.
My hardware is:
-5 budgetone 100
-2 handytone-286
-Asterisk server running on Pentium IV, RAM 1GB,
RedHat 8.0.
I've just upgraded Asterisk and modified
extensions.conf. My extensions.conf:
; extensions.conf
[globals]
;static=yes
EXTEN106=SIP/sip1
EXTEN107=SIP/sip2
EXTEN108=SIP/sip3
EXTEN109=SIP/sip4
EXTEN110=SIP/sip5
EXTEN111=SIP/sip6
EXTEN112=SIP/sip7
[telefonos]
include = todos
[todos]
exten = _1XX,1,NoOp()
exten = _1XX,2,Dial(${EXTEN${EXTEN}})
exten = _1XX,3,Hangup
exten = _1XX,4,Hangup
exten = _1XX,103,Hangup
exten = h,1,Hangup
exten = t,1,Hangup
My sip.conf
[general]
disallow=all
allow=alaw
;tos=lowdelay
bindaddr=0.0.0.0
nat=no
language=es
[sip1]
type=friend
secret=sip1
host=dynamic
defaultip=172.16.190.100
dtmfmode=rfc2833
context=telefonos
callerid=sip1 106
[sip2]
type=friend
secret=sip2
host=dynamic
defaultip=172.16.190.101
dtmfmode=rfc2833
context=telefonos
callerid=sip2 107
[sip3]
type=friend
secret=sip3
host=dynamic
defaultip=172.16.190.102
dtmfmode=rfc2833
context=telefonos
callerid=sip3 108
[sip4]
type=friend
secret=sip4
host=dynamic
defaultip=172.16.190.103
dtmfmode=rfc2833
context=telefonos
callerid=sip4 109
[sip5]
type=friend
secret=sip5
host=dynamic
defaultip=172.16.190.104
dtmfmode=rfc2833
context=telefonos
callerid=sip5 110
[sip6]
type=friend
secret=sip6
host=dynamic
defaultip=172.16.190.105
dtmfmode=rfc2833
context=telefonos
callerid=sip6 111
[sip7]
type=friend
secret=sip7
host=dynamic
defaultip=172.16.190.106
dtmfmode=rfc2833
context=telefonos
callerid=sip7
And the problem is still the same: Asterisk doesn't
detect the budgetone hangup.
The configuration of the Grandstream phones are:
-for sip1
http://tonidiego.webcindario.com/sip1.htm
-for sip2
http://tonidiego.webcindario.com/sip2.htm
-for sip3
http://tonidiego.webcindario.com/sip3.htm
-for sip4
http://tonidiego.webcindario.com/sip4.htm
-for sip5
http://tonidiego.webcindario.com/sip5.htm
-for sip6
http://tonidiego.webcindario.com/sip6.htm
-for sip7
http://tonidiego.webcindario.com/sip7.htm
Thanks in advance
OK. I solved the problem.
MUCHAS GRACIAS SERGIO.
We added the global parameter port=5060 in sip.conf.
That's all.
Everything is working OK.
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[Asterisk-Users] cdr_odbc with mysql on a remote server
I'm trying to add cdr_odbc.so to log my CDR data to a mysql DB. I've
managed to compile everything, and seem to almost be ready to head home.
I've added a small debug line to cdr_odbc.c as follows:
if((ODBC_res != SQL_SUCCESS) (ODBC_res !=
SQL_SUCCESS_WITH_INFO))
{
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
SQLConnect %d\n, ODBC_res);
SQLGetDiagRec(SQL_HANDLE_DBC, ODBC_con, 1, ODBC_stat,
ODBC_err, ODBC_msg, 100, ODBC_mlen);
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
Details: %s\n, ODBC_msg);
SQLFreeHandle(SQL_HANDLE_ENV, ODBC_env);
connected = 0;
return -1;
}
Lines marked with are lines I added. Here are the error messages I
get on the console:
asterisk*CLI load cdr_odbc.so
Loaded /usr/lib/asterisk/modules/cdr_odbc.so = (ODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_odbc.conf': == Parsing
'/etc/asterisk/cdr_odbc.conf': Found
2004-05-26 18:13:54 NOTICE[6151]: cdr_odbc.c:336 odbc_load_module:
cdr_odbc: Logging uniqueid
cdr_odbc: dsn is AsteriskCDR
cdr_odbc: username is asteriskcdr
cdr_odbc: password is [secret]
cdr_odbc: Error SQLConnect -1
cdr_odbc: Error Details: [MySQL][ODBC 3.51 Driver]Can't connect
to local MySQL server through socket '/tmp/mysql.sock' (2)
2004-05-26 18:13:54 ERROR[6151]: cdr_odbc.c:363 odbc_load_module:
cdr_odbc: Unable to connect to datasource: AsteriskCDR
cdr_odbc: Unable to connect to datasource: AsteriskCDR
So, the problem I am having is that the mysql odbc driver seems to want
to use a local socket, but I am not running mysql locally on the
asterisk machine. I want it to connect to a remote host.
I don't see anything in the source that indicates it can/should be able
to do this. Can someone either tell me it isn't possible, or I need to
hack the source, or it is already there and I am just blind...
Thanks,
Adam
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RE: [Asterisk-Users] Using Ser and Asterisk together
Hi,
It's possible, we're using a configuration like that.
1. Configure diferent sip listening ports for SER (/etc/ser/ser.cfg) and
Asterisk (/etc/asterisk/sip.conf).
2. Configure SER (/etc/ser/ser.cfg) for forwarding calls based in
destination. For example adding:
if (uri=~^sip:[EMAIL PROTECTED]) {
forward( 10.10.10.10, 5070 ); //Where local asterisk is
listening
break;
}
(Documentation in SER admin guide
http://www.iptel.org/ser/doc/seruser/seruser.html)
3. Configure Asterisk as PSTN gateway. I haven't experience in this point.
Good luck.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Aiden Chew
Enviado el: martes, 25 de mayo de 2004 9:21
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Using Ser and Asterisk together
Hi all,
I would like to know if it is possible to use asterisk and ser together in a
single computer system using ser as a sip proxy and forwarding any voice
call request to asterisk for calling into the pstn gateway. (or any other
alternative that is possible is also welcomed for suggestions). If it is
possible can someone kindly show me the necessary configuration files or
refer me to any page that can show me how to do it ? Thanks a lot in
advance.
Kevin
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Re: [Asterisk-Users] 11 instead of Star
On Tue, May 25, 2004 at 10:37:25AM -0500, Greg Blakely wrote: [...] + It's just as well that *8# isn't used for call pickup anymore. The # on the end really SHOULD mean end of dialing, and not have any other significance. Unfortunately, BT and GSM service codes give significance to # in the middle of the dialling sequence: *NN# - Enable service with code NN #NN# - Disable service *#NN# - Query status of service Or has this already been discussed to death? Possibly, but some of us are still arguing over the corpse :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More SIP channel information
Hi All
Does anyone know how I can get more information about an incoming SIP call
from a SIP proxy. Like FWD or any other SER proxy. My * box shows the
channel name as:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: 26 May 2004 10:22 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cdr_odbc with mysql on a remote server
I'm trying to add cdr_odbc.so to log my CDR data to a mysql DB. I've
managed to compile everything, and seem to almost be ready to
head home.
I've added a small debug line to cdr_odbc.c as follows:
if((ODBC_res != SQL_SUCCESS) (ODBC_res !=
SQL_SUCCESS_WITH_INFO))
{
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
SQLConnect %d\n, ODBC_res);
SQLGetDiagRec(SQL_HANDLE_DBC, ODBC_con, 1, ODBC_stat,
ODBC_err, ODBC_msg, 100, ODBC_mlen);
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
Details: %s\n, ODBC_msg);
SQLFreeHandle(SQL_HANDLE_ENV, ODBC_env);
connected = 0;
return -1;
}
Lines marked with are lines I added. Here are the error messages I
get on the console:
asterisk*CLI load cdr_odbc.so
Loaded /usr/lib/asterisk/modules/cdr_odbc.so = (ODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_odbc.conf': == Parsing
'/etc/asterisk/cdr_odbc.conf': Found
2004-05-26 18:13:54 NOTICE[6151]: cdr_odbc.c:336 odbc_load_module:
cdr_odbc: Logging uniqueid
cdr_odbc: dsn is AsteriskCDR
cdr_odbc: username is asteriskcdr
cdr_odbc: password is [secret]
cdr_odbc: Error SQLConnect -1
cdr_odbc: Error Details: [MySQL][ODBC 3.51
Driver]Can't connect
to local MySQL server through socket '/tmp/mysql.sock' (2)
2004-05-26 18:13:54 ERROR[6151]: cdr_odbc.c:363 odbc_load_module:
cdr_odbc: Unable to connect to datasource: AsteriskCDR
cdr_odbc: Unable to connect to datasource: AsteriskCDR
So, the problem I am having is that the mysql odbc driver
seems to want
to use a local socket, but I am not running mysql locally on the
asterisk machine. I want it to connect to a remote host.
I don't see anything in the source that indicates it
can/should be able
to do this. Can someone either tell me it isn't possible, or I need to
hack the source, or it is already there and I am just blind...
Thanks,
Adam
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[Asterisk-Users] Mor info about SIP channel
Hi all
Does nayone know how to get more specific info about an incoming SIP call
from a SIP proxy like FWD or any other SER proxy. All incmoing calls into my
* box from FWD and other SER proxies have the following channel name:
SIP/-081833b8 or something similar but with the same format (random)
Ie they are quite random. I would far rather have something like
SIP/fwd.pulver.comXX
Does anyone have any suggestions
Thankls in advance
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: 26 May 2004 10:22 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cdr_odbc with mysql on a remote server
I'm trying to add cdr_odbc.so to log my CDR data to a mysql DB. I've
managed to compile everything, and seem to almost be ready to
head home.
I've added a small debug line to cdr_odbc.c as follows:
if((ODBC_res != SQL_SUCCESS) (ODBC_res !=
SQL_SUCCESS_WITH_INFO))
{
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
SQLConnect %d\n, ODBC_res);
SQLGetDiagRec(SQL_HANDLE_DBC, ODBC_con, 1, ODBC_stat,
ODBC_err, ODBC_msg, 100, ODBC_mlen);
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
Details: %s\n, ODBC_msg);
SQLFreeHandle(SQL_HANDLE_ENV, ODBC_env);
connected = 0;
return -1;
}
Lines marked with are lines I added. Here are the error messages I
get on the console:
asterisk*CLI load cdr_odbc.so
Loaded /usr/lib/asterisk/modules/cdr_odbc.so = (ODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_odbc.conf': == Parsing
'/etc/asterisk/cdr_odbc.conf': Found
2004-05-26 18:13:54 NOTICE[6151]: cdr_odbc.c:336 odbc_load_module:
cdr_odbc: Logging uniqueid
cdr_odbc: dsn is AsteriskCDR
cdr_odbc: username is asteriskcdr
cdr_odbc: password is [secret]
cdr_odbc: Error SQLConnect -1
cdr_odbc: Error Details: [MySQL][ODBC 3.51
Driver]Can't connect
to local MySQL server through socket '/tmp/mysql.sock' (2)
2004-05-26 18:13:54 ERROR[6151]: cdr_odbc.c:363 odbc_load_module:
cdr_odbc: Unable to connect to datasource: AsteriskCDR
cdr_odbc: Unable to connect to datasource: AsteriskCDR
So, the problem I am having is that the mysql odbc driver
seems to want
to use a local socket, but I am not running mysql locally on the
asterisk machine. I want it to connect to a remote host.
I don't see anything in the source that indicates it
can/should be able
to do this. Can someone either tell me it isn't possible, or I need to
hack the source, or it is already there and I am just blind...
Thanks,
Adam
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[Asterisk-Users] act via Network or web?
Hello to all. I'm new to this list. I have one problem: I want test my extensions and configs (that is on phones ) without phone card.I want to know that how can i simulate Teh Call In to Ethernet (maybe the Web) and test my configs? thanks. regards. A-h.Ahmadi __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, recently wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list. You'll find it on http://lists.digium.com, which is the address where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Be a community member The Asterisk software growth is very much based on user contributions. That's really how we all pay for the software - and get revenue back. If you develop custom functionality, you can rest assured that there is someone out there that wants it, needs it and will be helped by it. Don't forget to contribute. Open Source is both giving and taking. The financial model behind it all is really cooperative in some way. As one member to the community said to a contractor: Hey, I'm paying you to deliver code to me, then I'm giving it away to the community. How did this happen? It's the Open Source business model. And if it didn't work, we wouldn't have a lot of the software platforms that we all use in our business systems - Linux, Apache, MySQL, PostgreSQL and Asterisk. ** Remember: It's Open Source, it's voluntary Asterisk.org is a Open Source project. This means you can't request help from people, demand new functions or support. However, there are many individuals and companies out there that are offering services based on Asterisk, from VoIP service providers to consultants all over the world. Of course, this is also part of Digium's business, so you have plenty of help if your willing to pay. Digium is to be found at
[Asterisk-Users] Interconnecting Asterisk with SER
Hi all, I need to connect an asterisk box with SER from an VoIP-Service Provider, any configuration example to do that are welcome. Thanks. Zouhair Echchelh Option-Service.fr --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.690 / Virus Database: 451 - Release Date: 22/05/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The materiel requirement for an asterisk with four T2 card
Hi all, I need to know if someone have an asterisk box with on, two, tree or fore T2 card, and which is the good materiel configuration to do that. Thanks. Zouhair Echchelh Option-Service.fr --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.690 / Virus Database: 451 - Release Date: 22/05/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phones
Tomica Crnek wrote: Hi everyone, I have to test few models of SIP IP phones with Asterisk. I have seen on voip-info.org that there are lot of phones that work ok with Asterisk. But, I want to ask for suggestion - which models are the best for Asterisk? We have a dozen or so Polycom IP 500's and IP 600's working great. I highly recommend them. -russ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem - Adtran TSU 600, t100p
On Tue, 25 May 2004, Bartosz Jozwiak wrote: Hello, I have just received Adtran TSU 600 with 24 FXS ports. I have installed sucessfuly T100P card. Sucessfully? Did you load the module for the card? Yes What does 'ztcfg -v' show? ast05:~# ztcfg -vvv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Slaves: 08) Channel 09: FXO Kewlstart (Default) (Slaves: 09) Channel 10: FXO Kewlstart (Default) (Slaves: 10) Channel 11: FXO Kewlstart (Default) (Slaves: 11) Channel 12: FXO Kewlstart (Default) (Slaves: 12) Channel 13: FXO Kewlstart (Default) (Slaves: 13) Channel 14: FXO Kewlstart (Default) (Slaves: 14) Channel 15: FXO Kewlstart (Default) (Slaves: 15) Channel 16: FXO Kewlstart (Default) (Slaves: 16) Channel 17: FXO Kewlstart (Default) (Slaves: 17) Channel 18: FXO Kewlstart (Default) (Slaves: 18) Channel 19: FXO Kewlstart (Default) (Slaves: 19) Channel 20: FXO Kewlstart (Default) (Slaves: 20) Channel 21: FXO Kewlstart (Default) (Slaves: 21) Channel 22: FXO Kewlstart (Default) (Slaves: 22) Channel 23: FXO Kewlstart (Default) (Slaves: 23) Channel 24: FXO Kewlstart (Default) (Slaves: 24) 24 channels configured. Is asterisk running? Does asterisk see the ports? (zap show channels) This is the following: *CLI zap show channels Chan Extension Context Language MusicOnHold 1defaultdefault 2defaultdefault 3default 4default 5default 6default 7default 8default 9default 10default 11default 12default 13default 14default 15default 16default 17default 18default 19default 20default 21default 22default 23default 24default Adtran is connected to t100p with crossover T1 cable. On T100P card I have a green light and on Adtran I do not get any errors or alarms. But I do not get dialtone on FXS ports. Adtran is configured: For Network Timing, fxs ports ore fxs_ls on Adtran. In zaptel.conf: snip Still having the same problem. I was palying around with span but it did not help. I have no idea anymore what could be wrong. Can some body be so kind and point me somewhere what I am doing worng. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] testing asterisk on FXS lines
On May 24, 2004, at 5:29 PM, Jay Milk wrote: For $49.99+SH I can sell you an FXO/FXS test-cable... just kidding. Use a regular RJ11 cable to connect one of your FXS ports to the FXO port you want to test, pick up another FXS and dial the extension... and you're promptly delivered to the [incoming] context. I test all my FXO configs using a Sipura FXS port to make it ring. I'd still like that $50 though :) Oh, this is a good idea. I guess I didn't think about being able to do that. Excellent! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Monday, May 24, 2004 3:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] testing asterisk on FXS lines On May 24, 2004, at 4:00 PM, Michael George wrote: I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to incoming so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Background(hello2) ; this is the file I need to test the playback of first And I do a restart. When I pickup one of the FXS handsets, though, I get this from asterisk (running with the -vvvc arg): Starting simple switch on 'Zap/1-1' and that is it. I know that the context is right because I put a hard-dial of 202 in there and when I dialed it, it would connect to that extension (Zap/2) and if I dialed anything else I would get fast busy. I have checked and the line right after the last exten above is another context marker. The asterisk output also shows the s extensions being loaded under the correct context when I do a reload after the restart (to see just the messages from the contexts being loaded). What am I missing to get the FXS lines, in the context incoming, to do the wait/answer/background? Thanks! For some reason, the s extension is not being executed for the FXS lines. I changed their default context back to internal and added exten = s,1,Background(hello2) to the internal context, thinking that when I pick up the handset I will get the hello2 audio file played as it waits for me to enter digits. But the audio file is not played... I must be missing an essential concept here... -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] testing asterisk on FXS lines
On May 24, 2004, at 8:46 PM, Jason Kawakami wrote: i always use the Goto application. seems to work quite well for testing those s extensions. exten = 2500,1,Goto(context,s,1) will take you to step 1 in the s extension in whatever context. Hmm, very interesting idea. Similar to putting misc. buttons on applications when testing esoteric functionality. Thanks for the tip! Jason Kawakami - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 5:20 PM Subject: Asterisk-Users digest, Vol 1 #3886 - 9 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Re: Making a SIP call (Eric Wieling) 2. RE: testing asterisk on FXS lines (Jay Milk) 3. SIP Authentication Problem (Chuck Ramirez) 4. RE: 2 Sip phones behind un-natted Asterisk (Chad Brown) 5. Re: extensions/sip from database? (Fran Boon) 6. Using Blacklist (Steven E. Frazier) 7. Asterisk connected to DataBase (pesb) 8. mpg123 (Simon Brown) 9. Re: Using Blacklist (Dorian Gray) --__--__-- Message: 1 Date: Mon, 24 May 2004 16:20:36 -0500 From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Making a SIP call Reply-To: [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am still having this problem of only capturing part of the IP address, I am currently checking into a possible hardware/software issue on the client side but was wondering if there are any setting I need to set on the asterisk server to allow an peer to peer call. I have set dtmfmode=inband. Is there anything else I need to set? dtmfmode=inband only works with the ulaw and alaw codecs. If you use any other codec you MUST use rfc2833 or info DTMF modes (set on the phone AND on Asterisk) --__--__-- Message: 2 From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] testing asterisk on FXS lines Date: Mon, 24 May 2004 16:29:39 -0500 Reply-To: [EMAIL PROTECTED] For $49.99+SH I can sell you an FXO/FXS test-cable... just kidding. Use a regular RJ11 cable to connect one of your FXS ports to the FXO port you want to test, pick up another FXS and dial the extension... and you're promptly delivered to the [incoming] context. I test all my FXO configs using a Sipura FXS port to make it ring. I'd still like that $50 though :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Monday, May 24, 2004 3:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] testing asterisk on FXS lines On May 24, 2004, at 4:00 PM, Michael George wrote: I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to incoming so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Background(hello2) ; this is the file I need to test the playback of first And I do a restart. When I pickup one of the FXS handsets, though, I get this from asterisk (running with the -vvvc arg): Starting simple switch on 'Zap/1-1' and that is it. I know that the context is right because I put a hard-dial of 202 in there and when I dialed it, it would connect to that extension (Zap/2) and if I dialed anything else I would get fast busy. I have checked and the line right after the last exten above is another context marker. The asterisk output also shows the s extensions being loaded under the correct context when I do a reload after the restart (to see just the messages from the contexts being loaded). What am I missing to get the FXS lines, in the context incoming, to do the wait/answer/background? Thanks! For some reason, the s extension is not being executed for the FXS lines. I changed their default context back to internal and added exten = s,1,Background(hello2) to the internal context, thinking that when I pick up the handset I will get the hello2 audio file played as it waits for me to enter digits. But the audio file is not played... I must be missing an essential concept here... -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 3 Date: Mon, 24 May 2004 14:25:00 -0700 (PDT) From: Chuck Ramirez [EMAIL PROTECTED] To: [EMAIL
Re: [Asterisk-Users] testing asterisk on FXS lines
On May 24, 2004, at 10:34 PM, Adam Goryachev wrote: Look in your zapata.conf (hmmm, or zaptel.conf I awlays get confused, the one in /etc/asterisk/zap???.conf) You need to add the line: immediate = yes This means as soon as you pick up the line, it will follow the 's' extension. This is what I happened upon myself at the end of the day when I posted the question. I like the cleanliness of the other two solutions better but your response allowed me to learn more about the workings of asterisk! (You will need this defined for your fxo interface as well later) That is what I would expect, but the sample files I have, as well as the one I am running, have immediate=no before the FXO or FXS lines and does not change it. I'm thinking that the fxs_ks signalling must override the immediate mode. Regards, Adam On Tue, 2004-05-25 at 10:46, Jason Kawakami wrote: i always use the Goto application. seems to work quite well for testing those s extensions. exten = 2500,1,Goto(context,s,1) will take you to step 1 in the s extension in whatever context. Jason Kawakami - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 5:20 PM Subject: Asterisk-Users digest, Vol 1 #3886 - 9 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Re: Making a SIP call (Eric Wieling) 2. RE: testing asterisk on FXS lines (Jay Milk) 3. SIP Authentication Problem (Chuck Ramirez) 4. RE: 2 Sip phones behind un-natted Asterisk (Chad Brown) 5. Re: extensions/sip from database? (Fran Boon) 6. Using Blacklist (Steven E. Frazier) 7. Asterisk connected to DataBase (pesb) 8. mpg123 (Simon Brown) 9. Re: Using Blacklist (Dorian Gray) --__--__-- Message: 1 Date: Mon, 24 May 2004 16:20:36 -0500 From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Making a SIP call Reply-To: [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am still having this problem of only capturing part of the IP address, I am currently checking into a possible hardware/software issue on the client side but was wondering if there are any setting I need to set on the asterisk server to allow an peer to peer call. I have set dtmfmode=inband. Is there anything else I need to set? dtmfmode=inband only works with the ulaw and alaw codecs. If you use any other codec you MUST use rfc2833 or info DTMF modes (set on the phone AND on Asterisk) --__--__-- Message: 2 From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] testing asterisk on FXS lines Date: Mon, 24 May 2004 16:29:39 -0500 Reply-To: [EMAIL PROTECTED] For $49.99+SH I can sell you an FXO/FXS test-cable... just kidding. Use a regular RJ11 cable to connect one of your FXS ports to the FXO port you want to test, pick up another FXS and dial the extension... and you're promptly delivered to the [incoming] context. I test all my FXO configs using a Sipura FXS port to make it ring. I'd still like that $50 though :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Monday, May 24, 2004 3:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] testing asterisk on FXS lines On May 24, 2004, at 4:00 PM, Michael George wrote: I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to incoming so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Background(hello2) ; this is the file I need to test the playback of first And I do a restart. When I pickup one of the FXS handsets, though, I get this from asterisk (running with the -vvvc arg): Starting simple switch on 'Zap/1-1' and that is it. I know that the context is right because I put a hard-dial of 202 in there and when I dialed it, it would connect to that extension (Zap/2) and if I dialed anything else I would get fast busy. I have checked and the line right after the last exten above is another context marker. The asterisk output also shows the s extensions being loaded under the correct context when I do a reload after the restart (to see just the messages from the contexts being loaded). What am I missing to get the FXS lines, in the context incoming, to do the wait/answer/background? Thanks! For some reason, the s extension is not being executed for the FXS
RE: [Asterisk-Users] SIP phones
My own personal opinion in order of preference: Snom 200 - excellent albeit a little too expensive Grandstream - excellent - absolutely excellentfor the price Cisco 7960 - wel - cool looking build like a tank but require some fiddling and WAY to expensive ZyXEL Prestige 2000W - too expensive and still extremely buggy - but cool when they get it fixed though - if ever Snom 100 - very poor mechanical quality Those are the ones I've been testing. I don't know if the new Snom 105 is better mechanical quality than the old Snom 100. If it is - it would raise to top of my list. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Tomica CrnekSent: 26 May 2004 16:18To: [EMAIL PROTECTED]Subject: [Asterisk-Users] SIP phones Hi everyone, I have to test few models of SIP IP phones with Asterisk. I have seen on voip-info.org that there are lot of phones that work ok with Asterisk. But, I want to ask for suggestion - which models are the best for Asterisk? I would appreciate if I can choose 4-5 models for test. Tomica Crnek
RE: [Asterisk-Users] spandsp hylafax asterisk and confusion
I too, would like to thank everyone for helping out. I did finally get everything working correctly. Terry [EMAIL PROTECTED] 5/25/2004 5:46:44 PM Thanks everyone for your responses. While these tips and tricks did infact help get asterisk compiled with the fax modules, it seems that * still craps out on the app_dtmftotext.c when you first start it. I can't seem to find a way to get rid of it. I'm not even totally sure it's required to send or receive faxes. If anyone has a step by step (more like, location by location) as a work around for that, I'd be all ears. I thought removing the lines in the Makefile for app_dtmftotext.c would be enough for it to be excluded, but apparently it's not. If it's this much of a pain to get the fax modules installed everytime I update from CVS, it makes me wonder if the $8/mo I pay to JFAX isn't worth it! =) Cheers, Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: [EMAIL PROTECTED] Web: http://www.akiva.com Phone: 760-710-3209 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wade J. Weppler Sent: Tuesday, May 25, 2004 2:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] spandsp hylafax asterisk and confusion Or just add /usr/local/lib to your /etc/ld.so.conf file. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: Tuesday, May 25, 2004 1:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp hylafax asterisk and confusion Brian D'Arcy wrote: ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory I copied the libspan* files from /usr/local/lib to /usr/lib and then asterisk started! klaus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?
Hi, WKH You are correct... No glare on a PRI Really? I followed some of the regular advice that's dispensed on this list and tried to RTFG. Interestingly it transpires that several hundred hits on Google seem to imply that you're both wrong: http://tinyurl.com/2vmrh ..and here are two PRI/Glare scenarios nicely documented by Intel (for their Linux stack before someone mentions M$): http://tinyurl.com/27her and http://tinyurl.com/27her Perhaps we are disagreeing over use of terminology rather than an event that can obviously occur. I understand why Scott raises the issue especially with the aggressive services that he supports using Asterisk. Perhaps Scott could use his call loop-back stress tester code to model the problem and let us know how Asterisk behaves in a test environment. (Although he might need two * machines, back to back, to recreate real circuit contention problems.) Just my 2c Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of W. Kevin Hunt Sent: 25 May 2004 23:27 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Glare condition - How well does asteriskhandle? You are correct... No glare on a PRI W. Kevin Hunt CCIE #11841 www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, May 25, 2004 3:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Glare condition - How well does asteriskhandle? On Tue, 2004-05-25 at 13:53, Scott Stingel wrote: Hi- I have an upcoming application that requires use of PRI channels that are primarily used for high-volume incoming traffic, but that are to be used for outbound calling as well. Of course, one option is to have dedicated outbound channels reserved, but this is an inefficient use of channel resources. Normally PBX's are designed to have the CPE yield to an incoming call if a particular channel is seized by both ends at the same time (a condition known as glare), but I'm wondering if anyone has real-world experience with asterisk to say how well this is handled. While I may be wrong, I don't think glare happens on PRI. The difference being that the call isn't sent over a channel until there had been communications on the D channel. This means a send and a receive. Glare would happen on a channelized T1 where it is possible for each end to try and seize the channel at the same time, since there isn't any out of band communications. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notify to external number
I'm not aware of any way of doing this currently, but this has made it to the planning board of Voicemail3... the timing for which is unfortunately undetermined at the moment. HTH, Ryan Thrash On May 25, 2004, at 11:14 AM, Bruce Komito wrote: When a user has voicemail, I would like * to call the user at a pre-determined number (internal or external) and play a message that the user has voicemail, and then give the user the option to login to voicemail and pick up the message. I know about the externnotify feature, but I don't see a way to use it to accomplish what I want. I've checked the archives, etc., but I don't see that anyone has ever done this. If you have, please respond. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Tim Robinson wrote: Tony - This sounds great. Are you monitoring the line constantly for the inbound caller ID or are you somehow detecting the polarity reversal? I keep rolling buffer of the last couple of seconds of the incoming audio, so when the ring is detected the chan_zap driver can grab this and feed it to the callerid processing routines. If it works and is stable, will you disclaim your code so that it will get merged into the main CVS? There should probably be a couple of settings in zapata.conf for the caller id coding scheme to be used for each card If it's necessary to assign copyright to digium then there's no problem doing that. At the moment there's a rather lame 'ukcallerid=yes' command... it needs something better certainly but there's plenty of time to get that stuff right. The current patches are at http://www.nodomain.org/asterisk/ since a lot of people here in UK have a line from BT and a cable co line, where the cable co either uses Bellcore after 1st ring, or V23 after 1st ring. So you need to be able to chose the method for each line. What a mess, eh? Ugh. V23 after first ring... It also matters of course if the cable co. has changed the wire data format - you might be able to grab the data but then not be able to make any sense of it.. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] dialing multiple extensions
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Roger
Sendt: 26. maj 2004 00:54
Til: [EMAIL PROTECTED]
Emne: Re: [Asterisk-Users] dialing multiple extensions
John Fraizer wrote:
It looks like your cellphone carrier is actually answering the call
before they ring your phone. In their switch, they probably have the
equiv of:
exten = your.cellphone.number,1,Answer()
exten = your.cellphone.number,2,Ringing
exten = your.cellphone.number,3,Dial(CELL/${EXTEN},20)
exten = your.cellphone.number,4,Voicemail(u${EXTEN})
exten = your.cellphone.number,104,Voicemail(b${EXTEN})
This is actually a serious no-no for them to be doing though. If that
is in fact what they are doing, anyone who dials long distance to your
cellphone will be paying to hear it ring, even if they hang up before
you answer or your voicemail answers the phone.
What they SHOULD be doing is more along these lines:
exten = your.cellphone.number,1,Dial(CELL/${EXTEN},20)
exten = your.cellphone.number,2,Voicemail(u${EXTEN})
exten = your.cellphone.number,102,Voicemail(b${EXTEN})
Good luck getting them to change this behavior though if they are
actually giving answer indication right off the bat.
Thanks for the reply - I have version cell phone service. I did a work
around and called my cell phone via IAX2 as opposed to the zaptel
channels. This works and all 3 extensions ring w/ no problem.
--
Rock River Internet Roger Grunkemeyer
202 W. State St, 8th Floor[EMAIL PROTECTED]
Rockford, IL 61101 815-968-9888 x102
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Re: [Asterisk-Users] zaprtc-for-2.6
Hi I just installed * via bri-stuff from junghanns.net. I also use Kernel 2.6.5 and it seems to work fine. I saw the directory zaprtc-for-2.6 coming with bri-stuff and noticed that it is not used by the install scripts. I have absolutely no idea what this software does. Can anybody clear me up? it looks like the dummy-timer-programm (like ztdummy or zaprtc) if you have no FXS or other digenum-card in your Asterisk. You need that for some MP3 / music-on-hold issues. these seems the version vor kernel 2.6 regards thorsten gehrig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR destination when user presses '#'
Mark Turner wrote:
If '#' is pressed during a call the CDR that is written at the end of
the call contains '#' in the dst / destination field rather than the
number that was originally called. How do I avoid losing that original
number so that I can use the CDR for billing?
I've tried not having a '#' target in extensions.conf and I've tried
calling ResetCDR(w) in the '#' target hoping that would cause a CDR to
be written with the original number but in both cases the CDR still
contains '#'.
Any ideas please?
Thanks,
Mark.
Mark,
I'm not sure if this helps, but I have experienced the same problem with
the special 'h', 't', 'T', etc. extensions. I worked around the problem
by saving the the original extension in a variable, then restoring it
using a Goto back to the original extension.
Perhaps something like this will work for you (I have not tested this):
exten = 5551212,1,SetVar(ORIG_EXTEN=${EXTEN})
exten = 5551212,2,Dial(Zap/555)
exten = 5551212,3,Hangup
exten = #,1,Goto(${ORIG_EXTEN},3)
Regards,
../fam
--
Frank A. Mandarino [EMAIL PROTECTED]
Spindrift Management, Toronto
416 642-3404
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Re: [Asterisk-Users] Can I do this ...
Il 06:52, mercoledì 26 maggio 2004, Shaun Ewing ha scritto: Try: exten = s,1,Playback(thanksforcalling) exten = s,2,Dial(SIP/SIP/1112|30|m) exten = s,3,Voicemail(uEXTEN) exten = s,4,Playback(vm-goodbye) That will answer and play back thanksforcalling.gsm, dial SIP/ and SIP/1112 with music. If not answered within 30 seconds, it will go to voicemail. You could also add a 103 line to be used if both extensions are busy (eg: voicemail(bEXTEN)). -Shaun Is it possible to answer with a message WHILE calling by dialling. in effect a sort of dial option A() [see show application dial] but for the calling party.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 11 instead of Star and how about #?
Il 10:34, mercoledì 26 maggio 2004, Peter Corlett ha scritto: On Tue, May 25, 2004 at 10:37:25AM -0500, Greg Blakely wrote: [...] + It's just as well that *8# isn't used for call pickup anymore. The # on the end really SHOULD mean end of dialing, and not have any other significance. Unfortunately, BT and GSM service codes give significance to # in the middle of the dialling sequence: *NN# - Enable service with code NN #NN# - Disable service *#NN# - Query status of service Or has this already been discussed to death? Possibly, but some of us are still arguing over the corpse :) Also here in italy. there is also another small problem, what happens if a called phone directory need to press the # to continue and # have a transfer mean for asterisk? It's possible to escape the # sequence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Tony, Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 26 May 2004 13:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 Tim Robinson wrote: Tony - This sounds great. Are you monitoring the line constantly for the inbound caller ID or are you somehow detecting the polarity reversal? I keep rolling buffer of the last couple of seconds of the incoming audio, so when the ring is detected the chan_zap driver can grab this and feed it to the callerid processing routines. If it works and is stable, will you disclaim your code so that it will get merged into the main CVS? There should probably be a couple of settings in zapata.conf for the caller id coding scheme to be used for each card If it's necessary to assign copyright to digium then there's no problem doing that. At the moment there's a rather lame 'ukcallerid=yes' command... it needs something better certainly but there's plenty of time to get that stuff right. The current patches are at http://www.nodomain.org/asterisk/ since a lot of people here in UK have a line from BT and a cable co line, where the cable co either uses Bellcore after 1st ring, or V23 after 1st ring. So you need to be able to chose the method for each line. What a mess, eh? Ugh. V23 after first ring... It also matters of course if the cable co. has changed the wire data format - you might be able to grab the data but then not be able to make any sense of it.. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 11 instead of Star and how about #?
Also here in italy. there is also another small problem, what happens if a called phone directory need to press the # to continue and # have a transfer mean for asterisk? Not just in Italy, this has been my biggest beef with using *any* normal DTMF to escape to asterisk. I realize that the need for such a thing but to have it hardcoded is a bad bad thing. With Zap channels you can always hookflash to drop back to asterisk, and with almost any SIP phone you have a transfer button so I'm not exactly sure why # exists (perhaps from cell phones, where hookflash doesn't work?) It's possible to escape the # sequence Not that I am aware of. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Presumably this can then be modified for DTMF caller ID by those in NL, Brazil etc? I will give it a go soon. Thanks Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: 26 May 2004 13:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 Tim Robinson wrote: Tony - This sounds great. Are you monitoring the line constantly for the inbound caller ID or are you somehow detecting the polarity reversal? I keep rolling buffer of the last couple of seconds of the incoming audio, so when the ring is detected the chan_zap driver can grab this and feed it to the callerid processing routines. If it works and is stable, will you disclaim your code so that it will get merged into the main CVS? There should probably be a couple of settings in zapata.conf for the caller id coding scheme to be used for each card If it's necessary to assign copyright to digium then there's no problem doing that. At the moment there's a rather lame 'ukcallerid=yes' command... it needs something better certainly but there's plenty of time to get that stuff right. The current patches are at http://www.nodomain.org/asterisk/ since a lot of people here in UK have a line from BT and a cable co line, where the cable co either uses Bellcore after 1st ring, or V23 after 1st ring. So you need to be able to chose the method for each line. What a mess, eh? Ugh. V23 after first ring... It also matters of course if the cable co. has changed the wire data format - you might be able to grab the data but then not be able to make any sense of it.. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
David J Carter wrote: Tony, Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? It's to allow the X100P to output the caller ID. My soldering skills just weren't up to the BT50 mod :) Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Robinson Tim-W10277 wrote: Presumably this can then be modified for DTMF caller ID by those in NL, Brazil etc? I will give it a go soon. I'd expect so. The zaptel mod just lets you grab what happened just before the ring... processing it is a relatively simple addition (no idea how you'd do DTMF though). Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound Distortion using IAX?
Hi All, At present calls over IAX2 (ilbc) are good but they suffer from occasional distortion. The strange thing is that the distortion can only be heard by the calling party and not the called party in 95% of cases. IAX2 is being used with trunking enabled, using the ztdummy module as a timing source. Bandwidth shouldn't be an issue as there is more than sufficient plus we use QoS (internally). For the record the asterisk box is running the latest version from cvs-head. Anyone got any suggestions? Does jitterbuffer work in cvs-head? Its currently disabled and I wondered if it would have any positive effect on call quality? Regards, Nathan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone got latest SIP image for Cisco 7960?
Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PostgreSQL
Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PostgreSQL
Dear, Check whether you have enable tcp/ip socket connection in your Postgres config. postgresql.conf, if yes, see whether u have respective user and password strategy 'trust'. Fabio Donaggio wrote: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone got latest SIP image for Cisco 7960?
If you were really qualified to be a tech member according to cisco you would know that you need CCO access for the images and that it would be illegal for someone to give them to you. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Wednesday, May 26, 2004 8:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Anyone got latest SIP image for Cisco 7960? Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PostgreSQL
On Wed, 2004-05-26 at 08:40, Fabio Donaggio wrote: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! Seems it told you pretty clear that it could not establish a connection to your postgresql server. I suggest you start by checking that it is running, that it is accessible via TCP/IP(maybe use nmap), and that you can connect to it with 'psql -h localhost'. Most likely you are not configured to answer TCP/IP. Look in /etc/postgresql/postgresql.conf for something like 'tcpip_socket = true' and 'port = 5432'. You may also need to have '-i' in POSTMASTER_OPTIONS under postmaster.conf. There is still the possibility then that you don't have the proper pg_hba.conf entry to allow connects in. Don't forget to restart postgres after your config changes so they take effect. I'm not sure how much it will allow to change when you kill -HUP it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr_odbc with mysql on a remote server
Its very clear why its not working. Visit www.voip-info.org for some
examples on how to set it up. If you didn't compile unixODBC and MyODBC
from src then you will need to do so. Installing via RPM or package has
proven in the past to not work.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Wednesday, May 26, 2004 3:22 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cdr_odbc with mysql on a remote server
I'm trying to add cdr_odbc.so to log my CDR data to a mysql DB. I've
managed to compile everything, and seem to almost be ready to head home.
I've added a small debug line to cdr_odbc.c as follows:
if((ODBC_res != SQL_SUCCESS) (ODBC_res !=
SQL_SUCCESS_WITH_INFO))
{
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
SQLConnect %d\n, ODBC_res);
SQLGetDiagRec(SQL_HANDLE_DBC, ODBC_con, 1, ODBC_stat,
ODBC_err, ODBC_msg, 100, ODBC_mlen);
if(option_verbose 10)
ast_verbose( VERBOSE_PREFIX_4 cdr_odbc: Error
Details: %s\n, ODBC_msg);
SQLFreeHandle(SQL_HANDLE_ENV, ODBC_env);
connected = 0;
return -1;
}
Lines marked with are lines I added. Here are the error messages I
get on the console:
asterisk*CLI load cdr_odbc.so
Loaded /usr/lib/asterisk/modules/cdr_odbc.so = (ODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_odbc.conf': == Parsing
'/etc/asterisk/cdr_odbc.conf': Found
2004-05-26 18:13:54 NOTICE[6151]: cdr_odbc.c:336 odbc_load_module:
cdr_odbc: Logging uniqueid
cdr_odbc: dsn is AsteriskCDR
cdr_odbc: username is asteriskcdr
cdr_odbc: password is [secret]
cdr_odbc: Error SQLConnect -1
cdr_odbc: Error Details: [MySQL][ODBC 3.51 Driver]Can't connect
to local MySQL server through socket '/tmp/mysql.sock' (2)
2004-05-26 18:13:54 ERROR[6151]: cdr_odbc.c:363 odbc_load_module:
cdr_odbc: Unable to connect to datasource: AsteriskCDR
cdr_odbc: Unable to connect to datasource: AsteriskCDR
So, the problem I am having is that the mysql odbc driver seems to want
to use a local socket, but I am not running mysql locally on the
asterisk machine. I want it to connect to a remote host.
I don't see anything in the source that indicates it can/should be able
to do this. Can someone either tell me it isn't possible, or I need to
hack the source, or it is already there and I am just blind...
Thanks,
Adam
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RE: [Asterisk-Users] Anyone got latest SIP image for Cisco 7960?
Sorry, that's illegal. You have to purchase the support options via Cisco that entitle you to software upgrades. It's $8.50 per phone through most retailers, but it takes 6-8 weeks for cisco to issue you a password. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Wednesday, May 26, 2004 8:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Anyone got latest SIP image for Cisco 7960? Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PostgreSQL
On Wed, 26 May 2004, Fabio Donaggio waxed: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! You need to make sure that PostgreSQL is running with the '-i' option for net connections OR that postgresql.conf contains the line 'tcpip_socket = 1' Are you sure that PostgreSQL is running on the same machine that Asterisk is running on ? How did you connect to the database to create the CDR table initially ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ?
On Wed, 26 May 2004, Florent Guiliani waxed: Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent, Yes, buy a computer and install Asterisk on it. C = computer T = Asterisk I = install -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Tony, The patches work great, picks up the BT callerid everytime. A really big thankyou! Chris - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 26, 2004 2:09 PM Subject: Re: [Asterisk-Users] Caller ID with BT CD50 David J Carter wrote: Tony, Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? It's to allow the X100P to output the caller ID. My soldering skills just weren't up to the BT50 mod :) Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PostgreSQL
Fabio, You need to enable tcp connectivity on psql. Wherever you configured the databases to live (/var/lib/pgsql/data on my machine) you'll find a file called postgres.conf. You need to read that and uncomment out the appropriate lines to get: tcpip_socket = true port = 5432 -brian Fabio Donaggio wrote: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bug or feature?
I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ?
The proper answer to the question is what do you have in mind? CTI is an often-misused and often-misunderstood term. Florent needs to provide more information on what he wants. For example, is he looking for call/data transfer (i.e. data follows voice)? Is he looking for a simple autodialer? Does he just want DNIS information to pop up on his computer? Is he looking for a full-blown VRU application with database integration? Smart fax server? Etc. Greg On Wed, 26 May 2004, Florent Guiliani waxed: Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent, Yes, buy a computer and install Asterisk on it. C = computer T = Asterisk I = install -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Chris Stenton wrote: Tony, The patches work great, picks up the BT callerid everytime. A really big thankyou! A big THANK YOU from me too!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 79XX converting
Shaun Ewing wrote: Try P0S30203 if you have it. I've found that when converting these phones to SIP, I have to load P0S30203 (which I put in OS79XX.txt). I then place the newer version in SIPDefault.cnf (eg: image_version: P0S3-06-2-00). That way new SIP phones will go to P0S30203 first to get the extended filename support and then to P0S3-06-2-00. I can confirm this.. It was madening to upgrade the phone from MCGP to SIP.. Especially the SIP 3.x series had problems if the filename was to long. or had hyphens in it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing MSN on zaphfc
Hi folks I'am looking for the right way to select the outgoing MSN on zaphfc for Euro-ISDN. I found some notes on the Wiki and I know it has to be done in the dialplan. Does anyone know the right way/code? THX -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ?
What do you expect out of the CTI? Screen pops with customer information? DID based on caller-id of the customer (e.g. default sales-rep)? All those things are possible with AGI scripts and * applications. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florent Guiliani Sent: Wednesday, May 26, 2004 2:58 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ? Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: RE: snom reporting busy when it shouldn't
I think it's obvious that there are two dialogs being set up (take a look at the call-id and from-tag). I think on the protocol level the behavior is ok, although not beautiful. But I assume that * should send only one INVITE. Maybe there is a second registration dangling and * is forking the request under a new call-id. Christian -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of nicolas Sent: Tuesday, May 25, 2004 7:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: RE: snom reporting busy when it shouldn't Christian, they send two INVITE. Below the sip debug of an dial without doing an answer before. (There are 2 Phones (100/200) and a sipgate registrar) INVITE from * RING from snom INVITE from * BUSY from snom CANCEL from * if you want i can send a sip debug from the call waiting indication matter but is like above, without the 2. INVITE: INVITE from * BUSY from snom CANCEL from * Hope you can help. nicolas Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK38407e55 From: 410137463 sip:[EMAIL PROTECTED];tag=as6d950cc4 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 25 May 2004 07:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 364 Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK38407e55 From: 410137463 sip:[EMAIL PROTECTED];tag=as6d950cc4 To: sip:[EMAIL PROTECTED];tag=l0ggp0vc1z Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=jet7pbic Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK120fd9bf From: 0410137463 sip:[EMAIL PROTECTED];tag=as2a7e2d4f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 25 May 2004 07:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 364 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5832dc08 From: 410137463 sip:[EMAIL PROTECTED];tag=as05d21741 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 25 May 2004 07:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 364 Sip read: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK120fd9bf From: 0410137463 sip:[EMAIL PROTECTED];tag=as2a7e2d4f To: sip:[EMAIL PROTECTED];tag=1se6rz4cq8 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=jet7pbic Content-Length: 0 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK120fd9bf From: 0410137463 sip:[EMAIL PROTECTED];tag=as2a7e2d4f To: sip:[EMAIL PROTECTED];tag=1se6rz4cq8 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5832dc08 From: 410137463 sip:[EMAIL PROTECTED];tag=as05d21741 To: sip:[EMAIL PROTECTED];tag=d2jhjs2gig Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=jet7pbic Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK1e64da5d From: 0410137463 sip:[EMAIL PROTECTED];tag=as4981b9a8 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 25 May 2004 07:49:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 364 Sip read: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK1e64da5d From: 0410137463 sip:[EMAIL PROTECTED];tag=as4981b9a8 To: sip:[EMAIL PROTECTED];tag=1pnv8t8wys Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=jet7pbic Content-Length: 0 Transmitting:CLI ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK1e64da5d From: 0410137463 sip:[EMAIL PROTECTED];tag=as4981b9a8 To: sip:[EMAIL PROTECTED];tag=1pnv8t8wys Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 Sip read: SIP/2.0 180
Re: [Asterisk-Users] bug or feature?
On Wed, 2004-05-26 at 09:33, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? Wait() shouldn't take dtmf. It does seem odd till you realize that that is why there is timeouts and a timeout extension. Basically, if you are awaiting information from a user, just end your current priority and allow the timeout in your context to work. As for gotoif(), if you are processing dtmf and you start into a extension, asterisk has determined a match existed and is following your instructions. Only when you hit a point where you aren't telling asterisk what to do should it start listening for DTMF again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMDI support in Asterisk ?
I can provide logins and dev env to an asterisk server with an SMDI serial connection to anyone willing to work on the SMDI bounty.we have looked into this and got the hardware setup but I dont have time to write the code.. Dave P [EMAIL PROTECTED] 5/25/2004 1:45:12 PM W. Kevin Hunt wrote: I'll add $1k to that bounty, and will put another bounty out for $3k for ss7 integration w/ full isup / imt support... John Bittner wrote: I am also looking for the SMDI support. I am willing to put up a bounty of 2K to get this writen. Anyone interested please email me off list. ok, I've added these to the Wiki: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SMDI http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SS7 Anyone who has more info on what needs doing should add info there, also any further contributions... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
On Wed, 26 May 2004, Maveric waxed: I've noticed that when i pass a wait in an exten = that it doesn't allow Are you talking about the Wait() application ? 'show application wait' for dtmf tone input. Also on another note i've noticed that when using Background() is what you want if you want to *wait* for DTMF. gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? GotoIf should execute a lot faster than your fingers can push buttons to send DTMF. Can you post the relevant section(s) of your extensions.conf ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 100 analog phones?? HOWTO?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steven Critchfield wrote: | I was also thinking that if I used more than one machine with TDMoE I |could potentially have better performance. | | | I still doen't understand why someone would do TDMoE when IAX provides | more flexibility. I agree. I'd stick with IAX or IAX2 trunking, from personal experience. ~ Using TDMoE caused my machines to become unstable, reboot, kernel oops/panic, etc. It was not a pleasant experience. And, you have to make sure that you start and stop the *'s together. I can't remember if I ever got that working or if it always trashed the other box when I tried to bring * down with TDMoE in place. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAtLR3uYsUrHkpYtARAgrjAJ0S5z0gd68jliCNFxt0qrevKarIiwCeJ3IY XW3yQMKSnRvDZJOLSVR8yQU= =gGrL -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] powered analog phone and adit 600 feedback problem
I'm using an adit 600 channel bank with some powered analog phones that have caller id features, and some unpowered analog phones. The unpowered phones sound great, but the powered phones have a problem with loud feedback if you speak to close to the handset mic. I've determined that its not the settings in asterisk or the digium cards by disconnecting the T1 line from adit 600 to asterisk box. I turned the gain down as far as I could. Any ideas? Some adit settings: pbz-cbank1 show 2:5 SLOT 2: Settings for FXS: channel 5: Type:VOICE Signaling: LS RxGain: -9dB TxGain: -9dB LineLength: SHORT pbz-cbank1 status 2:5 FXSRx AB Tx AB Signal=T1 Sig T1 TP ---- - -- - -- 2:5 01 01 LS = LS Traffic N ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_odbc with mysql on a remote server
Adam Goryachev wrote: So, the problem I am having is that the mysql odbc driver seems to want to use a local socket, but I am not running mysql locally on the asterisk machine. I want it to connect to a remote host. This is an ODBC issue, not an Asterisk issue. Check odbc.ini F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] D-Channel on span 1 up/down + frame slips with zaptelBRI
Tobias Jönsson wrote: I have installed two HFC PCI A-cards running zaphfc from bristuff-0.0.2, which seems to work quite fine, but I continously receive the messages D-Channel on span 1 up followed by D-Channel on span 1 down with a few seconds interval. Why is that? Bri intense debug log and configuration files below. Hej Tobias. By judging from Your domain and lastname I guess You live in vikingland ;) As I have learned from kapejod, Telia (swedish telco) tries to power down the D-Channel and then the driver wakes up the channel again with a poll. So it's just normal Ha en underbar dag! /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CTI (Computer-Telephony Integration) with Asterisk ?
An eGroupware integration would be great. nico Gregory Junker wrote: The proper answer to the question is what do you have in mind? CTI is an often-misused and often-misunderstood term. Florent needs to provide more information on what he wants. For example, is he looking for call/data transfer (i.e. data follows voice)? Is he looking for a simple autodialer? Does he just want DNIS information to pop up on his computer? Is he looking for a full-blown VRU application with database integration? Smart fax server? Etc. Greg On Wed, 26 May 2004, Florent Guiliani waxed: Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent, Yes, buy a computer and install Asterisk on it. C = computer T = Asterisk I = install -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PostgreSQL
Thaks to all!!! Now it works! Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tdm04b stopped taking inbound calls - todays cvs
Has anyone tried the current Head cvs with TDM04b (4-port fxo)? The card stopped answering inbound calls (no CLI indications whatsoever), although outbound pstn calls via the card work just fine. Kind of looks like one of the changes from yesterday (probably wcfxs.c) might be causing the problem. (Total new checkout, install, reboot, etc, result in the same no-answer condition.) Anyone else seeing this? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Tony, Are you going to submit the patches to the cvs head? Chris - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 26, 2004 4:15 PM Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Tony Hoyle [EMAIL PROTECTED] wrote: Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? It's to allow the X100P to output the caller ID. I can confirm that it works for me. I applied the patches, compiled and installed zaptel, compiled and installed Asterisk and it just worked. Well done. I'll abandon my attempt now. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 79XX converting
lists wrote: CAN SOMEONE PLEASE POST THIS CONVERT IN A HOWTO OR FAQ This works Small twist I had to use signed loads so it was P0S353 is the OS79XX.txt and the SIPDEFAULT.cnf has the current load that I wanted to go to. Every other Doc or person that replied there howto never worked it may have been that I was on 5.0(1.1) SCCP which is a signed load but it never gave me the sign load error in tftp or on the phone before. In your SIPDefault put # Image Version image_version: P0S30201 In you OS79XX.TXT put P0S30201 Make sure in your tftp root director you have a file 'P0S30201' and it should be have permissions of 755. Next do a tcpdump tcpdump host ip of phone and port 69 That way when the phone boots you'll have a listing of the filenames its trying to transfer. Remember once you upgrade past 5.0 you can't go back to a lower version. After 5.0 the SIP images are digitally signed. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x102 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: RE: RE: snom reporting busy when it shouldn't
Christian, I think it's obvious that there are two dialogs being set up (take a look at the call-id and from-tag). I think on the protocol level the behavior is ok, although not beautiful. Ok, i see one from 04101... and one from 4101... But I assume that * should send only one INVITE. Maybe there is a second registration dangling and * is forking the request under a new call-id. Hm what can i do with a dangling registration ? Can it be there is a problem with the modem.conf ? I have registered there are strange behaivor with it. * is makeing executions for capi AND for modem/i4l: -- creating pipe for PLCI=0x101 msn = 3709387 sent ALERT_REQ PLCI = 0x101 -- Executing Wait(CAPI[contr1/3709387]/10, 1) in new stack -- started pbx on channel (callgroup=2)! -- Executing Wait(Modem[i4l]/ttyI0, 1) in new stack -- Executing SetLanguage(CAPI[contr1/3709387]/10, de) in new stack -- Executing SetMusicOnHold(CAPI[contr1/3709387]/10, default) in new stack -- Executing DigitTimeout(CAPI[contr1/3709387]/10, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(CAPI[contr1/3709387]/10, 15) in new stack -- Set Response Timeout to 15 -- Executing Wait(CAPI[contr1/3709387]/10, 2) in new stack -- Executing SetLanguage(Modem[i4l]/ttyI0, de) in new stack -- Executing SetMusicOnHold(Modem[i4l]/ttyI0, default) in new stack -- Executing DigitTimeout(Modem[i4l]/ttyI0, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(Modem[i4l]/ttyI0, 15) in new stack -- Set Response Timeout to 15 -- Executing Wait(Modem[i4l]/ttyI0, 2) in new stack -- Executing Dial(CAPI[contr1/3709387]/10, SIP/100SIP/200|20|Ttr) in new stack -- Executing Dial(Modem[i4l]/ttyI0, SIP/100SIP/200|20|Ttr) in new stack -- Called 100 -- Called 100 -- Called 200 -- SIP/100-4790 is ringing -- Got SIP response 486 Busy Here back from 190.100.200.19 -- SIP/200-5942 is ringing -- Called 200 -- SIP/100-219a is busy -- Got SIP response 486 Busy Here back from 190.100.200.18 -- SIP/200-80fc is busy == Everyone is busy at this time May 26 17:10:25 NOTICE[688153]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to SLINR -- Executing Wait(Modem[i4l]/ttyI0, 2) in new stack -- SIP/100-4790 is ringing -- SIP/200-5942 is ringing -- SIP/100-4790 is ringing -- SIP/200-5942 is ringing -- Executing VoiceMail(Modem[i4l]/ttyI0, u100) in new stack == Spawn extension (default, s, 7) exited non-zero on 'CAPI[contr1/3709387]/10' -- CAPI Hangingup -- removed pipe for PLCI = 0x101 -- Playing 'voicemail/default/100/unavail' (language 'de') == Spawn extension (default, s, 9) exited non-zero on 'Modem[i4l]/ttyI0' -- Hungup 'Modem[i4l]/ttyI0' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outgoing MSN on zaphfc
Hi Thomas!
you have to set the MSN this way for zaphfc when you use the dial command:
exten = _0Z.,5,Dial(CAPI/MyMSN:${EXTEN},90,mT)
Of course you have to set MyMSN to your MSN.
Regards,
Julian
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[Asterisk-Users] Newbie Here
Hello All Bit of a daft question but i am new to asterisk just got the box installed with 2 e1's . Are there any good examples of config ? ie how to make a phone number at the e1 route to a phone's ip etc which config files should i be looking at and how ? Thanks from a daft bloke Best Regards Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Can confirm it works with Generic X101P *BIG* Thank you :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 26 May 2004 16:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Tony Hoyle [EMAIL PROTECTED] wrote: Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? It's to allow the X100P to output the caller ID. I can confirm that it works for me. I applied the patches, compiled and installed zaptel, compiled and installed Asterisk and it just worked. Well done. I'll abandon my attempt now. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ?
Jay Milk wrote : What do you expect out of the CTI? Screen pops with customer information? DID based on caller-id of the customer (e.g. default sales-rep)? All those things are possible with AGI scripts and * applications. I'm looking for screen pops with customer information, is it possible and easy to do with asterisk? Florent, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: tdm04b stopped taking inbound calls - todays cvs
Has anyone tried the current Head cvs with TDM04b (4-port fxo)? The card stopped answering inbound calls (no CLI indications whatsoever), although outbound pstn calls via the card work just fine. Kind of looks like one of the changes from yesterday (probably wcfxs.c) might be causing the problem. (Total new checkout, install, reboot, etc, result in the same no-answer condition.) Anyone else seeing this? Responding to my earlier post above, backing out the zaptel changes from yesterday (cvs -D 2004...) fixed the problem. The tdm04b now answers inbound calls correctly. The backout replaced wcfxs.c and zonedata.c (which Mark updated yesterday for what appears to be some other fxs issues unrelated to fxo use). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rejecting Calls (SIT Tone/Invalid) Across PRI
Question regarding a callback application: I have a client who wants to allow callers to dial a DID which connects over a PRI to Asterisk. Asterisk will be analyzing the ANI data from each call to that DID and if it recognizes the ANI, it needs to effectively return an Invalid Number or Not Found some-such message across the PRI to prevent the user from being billed. Being a callback app, the system will then queue a callback to the party. The question is... How do I make Asterisk tell the network to disregard the billing and indicate to the caller that the callback has been scheduled? Is this possible? Is this legal? My fallback is simply to not answer the call if the ANI is recognized, then queue the callback when the incoming attempt terminates (i.e. the caller hangs up). Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Chris Stenton wrote: Tony, Are you going to submit the patches to the cvs head? http://bugs.digium.com/bug_view_page.php?bug_id=0001719 Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Chris Stenton [EMAIL PROTECTED] wrote: Tony, Are you going to submit the patches to the cvs head? He has published his patches, but probably doesn't have commit access to the Asterisk CVS archive (correct me if I'm wrong). The people who do have CVS commit access can pick up the published patch files and apply them if they want. I would suggest that someone with commit access should examine the patches and make a decision, as the changes appear to work very well. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp hylafax asterisk and confusion
I appreciate all the support SpanDSP is getting on this list, and the work that went into developing it. It's nice to see that Asterisk actually supports sending and receiving faxes now. One thing I was expecting to see here is that HylaFax works with SpanDSP, but I haven't seen any documentation for this. Is it possible to use SpanDSP in conjunction with HylaFax to route the printing and/or emailing of faxes. -- Christopher Lewis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 79XX converting
lists wrote: Humm that SCCP to start sorry I went up to a signed load on the sccp NP using my CCM's but I can't get the phone to load a SIP load. I am currently trying 7.1 as per cisco's paper http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guide s09186a008022a968.html#wp1048832 Still no go now I get a CTLSEP error on my solarwinds TFTP app. I read up on this some at cisco but it claims I just need OS79XX.txt -P00307100 (sip 7.1) and the SIPMAC but still no go. I did even CP the name from the load to the OS79XX.txt so I don't have a 0ZERO to O'OHHH' problem. If you're running a .sbn, you have to run .sbn from now on. You can't go back to non-signed binaries. Make sure that you have the appropriate .sbn SIP image on your tftp server. I've not had any problems at all upgrading *many* 7960's but, I suppose I could just be lucky. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
On Wednesday 26 May 2004 17:07, Karl Dyson wrote: Can confirm it works with Generic X101P *BIG* Thank you :) I can confirm it works with my generic X100P (at least I think that's what it is :) ). The full callerID is put into my database, so I know it's receiving the complete CID. The phone only seems to get sent the first 8 digit's - I'm sure this is something in my configs, but I've not had chance to look into it yet. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forwarding and record
Hi! my problem is to forwarding a call to a SIP phone and record the call at the same time. How can I do? This should help you to solve your problem: http://www.voip-info.org/wiki-Monitor+setup+sample Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 100 analog phones?? HOWTO?
Hi Petr! I use about 300 IP phone combination Welltech LP101, welltech LP102 welltech3502-8, Cisco 7905 and Cisco 7960 Would you be able to write a short report about your findings with the Welltech phones and post it to this list? I assume you are operating them in SIP mode. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Authentication Problem
Hi! Searchingat google, I found out that someone had a very similar problem to mine and posted it here under thesubject "Outgoing calls to SIP provider". Unfortunately I couldn't find how the problem was solved - if it was. Is it an Asterisk's bug? Have I done something wrong? Thanks, Chuck RamirezChuck Ramirez [EMAIL PROTECTED] wrote: I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials. Here is part of the content of sip.conf. [general]port = 5061bindaddr = *.IPcontext = invalidcalls ;This account is used for inbound and outbound callsregister = myuser:[EMAIL PROTECTED]/999 [mydomain]type=peerhost=myproxycontext=sipusername=myusersecret=mypassfromuser=myuserfromdomain=mydomain [user1]type=friendhost=dynamicdefaultip=default.IPusername=user1secret=secret1dtmfmode=rfc2833context=userscallerid="User 1"nat=yes Here is part of the content of extensions.conf. ;This part is working fine[sip]exten = 999,1,Dial(SIP/user1,,tr) [users]exten = _8.,1,Dial,SIP/[EMAIL PROTECTED],tr When I dial the number 812345 from my SIP Phone, this is the message sequencePhone - Asterisk: INVITE sip:[EMAIL PROTECTED] SIP/2.0Asterisk - Phone: SIP/2.0 407 Proxy Authentication RequiredPhone - Asterisk: ACK sip:[EMAIL PROTECTED] SIP/2.0Phone - Asterisk: INVITE sip:[EMAIL PROTECTED] SIP/2.0 (with authentication header)Asterisk - Phone: SIP/2.0 100 TryingAsterisk - Proxy: INVITE sip:[EMAIL PROTECTED] SIP/2.0Proxy - Asterisk: SIP/2.0 401 UnauthorizedAsterisk - Proxy: ACK sip:[EMAIL PROTECTED] SIP/2.0 The next message I would expect is another INVITE from * to the proxy with the authentication header.Why * hasn't send it? Can someone give me a help? Thanks in advance Chuck Ramirez Do you Yahoo!?Friends. Fun. Try the all-new Yahoo! Messenger Do you Yahoo!?Friends. Fun. Try the all-new Yahoo! Messenger
[Asterisk-Users] Monitoring Calls
I'm trying to set up basic monitoring for a specific extension (5004) to
record all outgoing and incoming calls and save them as WAV files. I've
set this in the extensions.conf file:
exten = 5004,1,Answer
exten = 5004,2,Wait,1
exten =
5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA
LLERIDNUM})
exten = 5004,4,Monitor,wav|${CALLFILENAME}
But it doesn't seem to work. Any guidance would be appreciated. I am
fairly new at Asterisk so my apologies if it is a really simple answer.
Thanks!
Mark
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Re: [Asterisk-Users] outgoing MSN on zaphfc
On Wed, May 26, 2004 at 05:54:52PM +0200, Julian Pawlowski wrote:
Hi Thomas!
you have to set the MSN this way for zaphfc when you use the dial command:
exten = _0Z.,5,Dial(CAPI/MyMSN:${EXTEN},90,mT)
zap=capi???
For Capi its clear but...
Maybe my mistake but I have zaphfc in TE Mode connected to TelCo and like
to use different MSN via the dialplan.
It could be that there is a code like *55(MY_MSN)# to prefix the call.
But I'am not shure if euroISDN has such thing.
Perhaps I'am totaly wrong??
It works with i4l, capi why not having the choice with zaphfc? :)
Of course you have to set MyMSN to your MSN.
Regards,
Julian
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--
Tho/\/\as
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[Asterisk-Users] tieline digit timeout
I'm connecting to an NEC t1 card via t100p (working great so far!) however I'm having problems dialing from the NEC system to an asterisk extension (sip-grandstream). If I hit the trunck line and dial REAL quick 103 I get the sip extension ringing; if I don't I get an invalad selection message from asterisk - and I can see on the console only one or two digits arrived. How can I globally make asterisk wait for several seconds on that t100p? Thanks, t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telus: Overseas calling
Paul Crick wrote: The question now is: how do I tell Asterisk to send everything starting with 011 as unknown numbering plan? You can use the pridialplan=unknown option in zapata.conf but that will then apply to everything Yeah, so setting pridialplan = unknown *and* changing switchtype = national to switchtype = 5ess did the trick. Thanks. -Markus -- Markus Mayer Calltrex Corporation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The materiel requirement for an asterisk with four T2 card
Hi! I need to know if someone have an asterisk box with on, two, tree or fore T2 card, and which is the good materiel configuration to do that. Have a look: http://www.voip-info.org/wiki-Asterisk+hardware+recommendations Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitoring Calls
On 26-May-04, at 12:02 PM, Mark Zawodny wrote:
I'm trying to set up basic monitoring for a specific extension (5004)
to
record all outgoing and incoming calls and save them as WAV files.
I've
set this in the extensions.conf file:
exten = 5004,1,Answer
exten = 5004,2,Wait,1
exten =
5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-
${CA
LLERIDNUM})
exten = 5004,4,Monitor,wav|${CALLFILENAME}
I use a slightly different syntax:
exten = 5004,4,Monitor(wav,${CALLFILENAME})
But it doesn't seem to work. Any guidance would be appreciated. I am
fairly new at Asterisk so my apologies if it is a really simple answer.
Thanks!
Mark
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Re: [Asterisk-Users] tieline digit timeout
On Wed, 2004-05-26 at 13:11, Tony wrote: I'm connecting to an NEC t1 card via t100p (working great so far!) however I'm having problems dialing from the NEC system to an asterisk extension (sip-grandstream). If I hit the trunck line and dial REAL quick 103 I get the sip extension ringing; if I don't I get an invalad selection message from asterisk - and I can see on the console only one or two digits arrived. How can I globally make asterisk wait for several seconds on that t100p? You don't want it to wait several seconds, you want it to only wait the amount of time to know what it is you keyed. My guess is you have some form of wildcard matching going on and you are getting caught early. Provide some of your configs for real help. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outgoing MSN on zaphfc
Hi,
like any other zaptel device, zaphfc uses the callerid from the
originating channel. If you want to override that callerid use:
exten = _X.,1,SetCallerID(MyMSN)
If you want to restrict the outgoing callerid (CLIR) make sure
you have usecallingpres=yes in zapata.conf and use:
exten = _X.,1,CallingPres(32)
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/
Am Mi, 2004-05-26 um 20.04 schrieb Thomas Niesel:
On Wed, May 26, 2004 at 05:54:52PM +0200, Julian Pawlowski wrote:
Hi Thomas!
you have to set the MSN this way for zaphfc when you use the dial command:
exten = _0Z.,5,Dial(CAPI/MyMSN:${EXTEN},90,mT)
zap=capi???
For Capi its clear but...
Maybe my mistake but I have zaphfc in TE Mode connected to TelCo and like
to use different MSN via the dialplan.
It could be that there is a code like *55(MY_MSN)# to prefix the call.
But I'am not shure if euroISDN has such thing.
Perhaps I'am totaly wrong??
It works with i4l, capi why not having the choice with zaphfc? :)
Of course you have to set MyMSN to your MSN.
Regards,
Julian
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RE: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ?
easy is such a subjective word. If you're a car-mechanic, it'll be incredibly difficult, if not impossible. If you're a linux-savvy C-programmer and you know the schema for your customer DB well, it should be trivial. You calling plan could... 1. Identify which extension(s) to ring 2. Run a custom app NotifyExt which sends caller-id to computers associated to certain extensions (or does a broadcast of extension/caller-id info) On your workstations, you'd have a program running that receives those notifications, or picks them out of the stream of broadcasts, populates customer information from you DB and pops it on the screen. So, from a management standpoint, this application is trivial -- all the pieces are readily available, you just need to put them together. Risks are access to your customer DB and your network topology. Security would be costly to implement, a broad-cast solution would be fairly easy to do. FWIW, I'm finishing my (home) Asterisk installation, Version 1.0, at the current time, and I have plans for implementing an IP based CDR (which displays current line usage in real time) as well as some CID based stuff, like standardizing incoming CID information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florent Guiliani Sent: Wednesday, May 26, 2004 11:13 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ? Jay Milk wrote : What do you expect out of the CTI? Screen pops with customer information? DID based on caller-id of the customer (e.g. default sales-rep)? All those things are possible with AGI scripts and * applications. I'm looking for screen pops with customer information, is it possible and easy to do with asterisk? Florent, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
The full callerID is put into my database, so I know it's receiving the complete CID. The phone only seems to get sent the first 8 digit's - I'm sure this is something in my configs, but I've not had chance to look into it yet. It looks like my missing digit problems are down to the dect phone I have connected to my handytone ata-286. When i have my Binatone dect connected, I only get the first 8 digits, if I connect my panasonic dect then I see all the digits - looks like I need a different dect phone :( Any ways, It looks like the patch works perfectly to me. It also works fine on my Telewest (Eurobell). Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] warning message (sound card) - when I run asterisk!!!
All,
After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:
chan_oss.so] = (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
== Registered channel type 'Console' (OSS Console
Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238
sound_thread: Read error on sound device: Resource
temporarily unavailable
[chan_phone.so] = (Linux Telephony API Support)
== Parsing '/etc/asterisk/phone.conf': Found
== Registered channel type 'Phone' (Standard Linux
Telephony API Driver)
My sound card information:
Vendor : Intel Corp.
Model : 82801CA/CAM AC'97 Audio Controller
Module : i810_audio
After running 'dial' command under the asterisk
prompt, I got the following message without any sound.
*CLI -- Executing Wait("OSS/dsp", "1") in new
stack
-- Executing Answer("OSS/dsp", "") in new stack
Console call has been answered
-- Executing DigitTimeout("OSS/dsp", "5") in new
stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("OSS/dsp", "10") in
new stack
-- Set Response Timeout to 10
-- Executing BackGround("OSS/dsp",
"demo-congrats") in new stack
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408
soundcard_setinput: Unable to re-open DSP device:
Device or resource busy
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567
oss_write: Unable to set device to input mode
May 26 00:40:55 WARNING[-1221268560]: file.c:537
ast_readaudio_callback: Failed to write frame
-- Playing 'demo-congrats' (language 'en')
== Spawn extension (local, s, 5) exited non-zero on
'OSS/dsp'
Is there anyone can give me any hints or help?
Thanks,
Neo
[Asterisk-Users] Sipura stun settings
I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk server and STUN server are outside the firewall on a public network. I would like the Sipuras to be able to reinvite, so I set canreinvite=yes in my sip.conf, and set the STUN server under the SIP tab in the Sipuras. However, I am not able to hear the other caller (the Sipura is not recieving RTP packets, it is sending just fine). Am I missing something on the Sipura config? I am not sure what all of the VIA options mean, and which ones I should use. Cant find any good info out there, can someone hrer help me out? Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Glare condition - How well does asterisk handle?
W. Kevin Hunt [EMAIL PROTECTED] wrote: __ Well that certainly seems to say there can be glare, but a read of the q.931 protocal stack seems to suport that it's not possible. If I were a leser man, I could say a snide remark about that if you click the home icon, it is apparent the document is referring to a driver and s/w for the Microsoft operating system to implement the PRI protocol ;) Which would be unfair, because Dialogic implement the pri protocols on the card, with a 486 and a heap of dsps on the one I have. Microsoft don't get a look in. This is a blessing and a curse. The good points are that they can get the card authorised for various telcos without reference to the os or cpu. Also it means that as a developer you are isolated from the os (somewhat). Also you can use it in an underpowered box, my E1 card is in a 180Mhz pentium pro! The downside is that the card is big and runs _hot_. Also Asterisk doesnt like it because it wants to get closer to the metal. On which subject, has anyone else got time to work with me on a chan_dialogicGC ? It looks do-able but I am ignorant of how asterisk does threading. W. Kevin Hunt CCIE #11841 www.huntbrothers.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk / SER or both
Hi, I am looking to implement a system to use as a prepaid service. I am aware that asterisk could do this with app_prepaid. However I am not sure if this is the best solution. Does anyone know if SER has a simmillar solution. Would I be right in assuming that SER as a SIP server is more scalable and can handle with more registerations/users ? Alternatively would it be a good idea to use SER as the SIP server and use asterisks as the PSTN gateway amd to manage the prepaid billing (given that only off net calls will be considered chargeable) Your commments and feedback will be greatly appreciated. Umar Sear ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Dev] SPAM MESSAGE - [Asterisk-Users] warning message (sound card) - when I run asterisk!!!
Hi Neo,
Your sound card is not well configured. Try to find
the right driver and load the correct module for it. ALSA (www.alsa-project.org/)
may help for this.
Hope this can help
Lamine
- Original Message -
From:
Neo Jia
To: [EMAIL PROTECTED] ; [EMAIL PROTECTED]
Sent: Wednesday, May 26, 2004 7:06
PM
Subject: [Asterisk-Dev] SPAM MESSAGE -
[Asterisk-Users] warning message (sound card) - when I run asterisk!!!
All,
After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:
chan_oss.so] = (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
== Registered channel type 'Console' (OSS Console
Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238
sound_thread: Read error on sound device: Resource
temporarily unavailable
[chan_phone.so] = (Linux Telephony API Support)
== Parsing '/etc/asterisk/phone.conf': Found
== Registered channel type 'Phone' (Standard Linux
Telephony API Driver)
My sound card information:
Vendor : Intel Corp.
Model : 82801CA/CAM AC'97 Audio Controller
Module : i810_audio
After running 'dial' command under the asterisk
prompt, I got the following message without any sound.
*CLI -- Executing Wait("OSS/dsp", "1") in new
stack
-- Executing Answer("OSS/dsp", "") in new stack
Console call has been answered
-- Executing DigitTimeout("OSS/dsp", "5") in new
stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("OSS/dsp", "10") in
new stack
-- Set Response Timeout to 10
-- Executing BackGround("OSS/dsp",
"demo-congrats") in new stack
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408
soundcard_setinput: Unable to re-open DSP device:
Device or resource busy
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567
oss_write: Unable to set device to input mode
May 26 00:40:55 WARNING[-1221268560]: file.c:537
ast_readaudio_callback: Failed to write frame
-- Playing 'demo-congrats' (language 'en')
== Spawn extension (local, s, 5) exited non-zero on
'OSS/dsp'
Is there anyone can give me any hints or help?
Thanks,
Neo
Re: [Asterisk-Users] bug or feature?
At 08:08 AM 5/26/2004, you wrote:
On Wed, 26 May 2004, Maveric waxed:
I've noticed that when i pass a wait in an exten = that it doesn't allow
Are you talking about the Wait() application ?
'show application wait'
for dtmf tone input. Also on another note i've noticed that when using
Background() is what you want if you want to *wait* for DTMF.
gotoif it will also cut the dtmf tones and drop the first part if the
gotoif is hit in the middle of input. Anybody else seen this or have this
problem?
GotoIf should execute a lot faster than your fingers can
push buttons to send DTMF. Can you post the relevant
section(s) of your extensions.conf ?
--Chris
--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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[easynews]
exten = s,1,SetVar,COUNTER=0;
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,BackGround(${SOUNDSDIR}/thank-you-for-calling-easynews)
exten = s,6,BackGround(${SOUNDSDIR}/new-direct-dial-number)
exten = s,7,BackGround(for-billing)
exten = s,8,BackGround(vm-press)
exten = s,9,BackGround(digits/1)
exten = s,10,BackGround(${SOUNDSDIR}/new-customer-signups)
exten = s,11,BackGround(vm-press)
exten = s,12,BackGround(digits/2)
exten = s,13,BackGround(for-tech-support)
exten = s,14,BackGround(vm-press)
exten = s,15,BackGround(digits/3)
exten = s,16,BackGround(${SOUNDSDIR}/if-you-know-your-partys-extension)
exten = s,17,BackGround(vm-press)
exten = s,18,BackGround(digits/5)
exten = s,19,SetVar,COUNTER=$[${COUNTER} + 1];
exten = s,20,GotoIf,$[${COUNTER} 4]?23:21
exten = s,21,Playback(vm-goodbye)
exten = s,22,Hangup
exten = s,23,Wait(0)
exten = 1,1,Goto(from-sip,1500,1)
exten = 2,1,Goto(from-sip,1505,1)
exten = 3,1,Goto(from-sip,1510,1)
exten = 5,1,Goto(ext-dial,s,1)
exten = 0,1,Goto(from-sip,1550,1)
exten = t,1,Goto(s,5)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,5)
It was much different before but this is how i worked around it. Also i
was calling Background before the wait but i think the wait should still
allow input.
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[Asterisk-Users] RE: Monitoring Calls
Unfortunately, that doesn't seem to work either. Any other suggestions?
Is there anything else that you would need to see (like the whole
extensions.conf file)? Your help is greatly appreciated!
Thanks!
Mark
From: Ryan Courtnage [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Monitoring Calls
Date: Wed, 26 May 2004 12:34:02 -0600
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
On 26-May-04, at 12:02 PM, Mark Zawodny wrote:
I'm trying to set up basic monitoring for a specific extension (5004)
to
record all outgoing and incoming calls and save them as WAV files.
I've
set this in the extensions.conf file:
exten = 5004,1,Answer
exten = 5004,2,Wait,1
exten =
5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-
${CA
LLERIDNUM})
exten = 5004,4,Monitor,wav|${CALLFILENAME}
I use a slightly different syntax:
exten = 5004,4,Monitor(wav,${CALLFILENAME})
But it doesn't seem to work. Any guidance would be appreciated. I am
fairly new at Asterisk so my apologies if it is a really simple
answer.
Thanks!
Mark
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-Original Message-
From: Mark Zawodny [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 26, 2004 2:02 PM
To: '[EMAIL PROTECTED]'
Subject: Monitoring Calls
I'm trying to set up basic monitoring for a specific extension (5004) to
record all outgoing and incoming calls and save them as WAV files. I've
set this in the extensions.conf file:
exten = 5004,1,Answer
exten = 5004,2,Wait,1
exten =
5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA
LLERIDNUM})
exten = 5004,4,Monitor,wav|${CALLFILENAME}
But it doesn't seem to work. Any guidance would be appreciated. I am
fairly new at Asterisk so my apologies if it is a really simple answer.
Thanks!
Mark
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