Re: [asterisk-users] Asterisk fail2ban filters - show us yours
Hi,
I Have added this line for asterisk 1.8 (i have allowguest=yes and
context=default in sip.conf):
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected because
extension not found in context 'default'.
Em 29-12-2011 13:03, Patrick Lists escreveu:
Hi,
In the thread Interesting attack tonight fail2ban them Bruce B mentioned
it would be nice to have input from the Community to come up with the best
set of fail2ban filters. That's a great idea. So let's start with Bruce's
filters (thanks!) and take it from there. Anyone have any improvements and/or
additions? Apologies for the line wrap. No idea how to prevent that in
Thunderbird. The filters are also at http://pastebin.com/6T9M1W3F
Not sure but it may be possible that logging has changed between Asterisk
1.4, 1.6, 1.8 and 10 so please mention the asterisk version with your filters.
For Asterisk 1.8:
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong
password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No
matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Device
does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer
is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')
There are 2 lines that I have which are not in this list:
NOTICE.* .*: Registration from '.*' failed for 'HOST' - ACL error
(permit/deny)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
How about those (no idea for which Asterisk version they are)?
Regards,
Patrick
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Re: [asterisk-users] Skype for Asterisk: Public Beta available
I have problems with it... [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler: Found license 'XX' providing 1 concurrent calls [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype For Asterisk Host-ID: X [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1320 sfa_startup: Found a total of 1 Skype For Asterisk licenses [Jul 30 14:34:21] WARNING[30613]: core.cpp:286 kill_skypewatcher: sending SIGTERM to 30614 failed with No such process *CLI [Jul 30 14:34:27] ERROR[30529]: core.cpp:1551 sfa_startup: Skype engine failed to start. [Jul 30 14:34:27] ERROR[30529]: chan_skype.c:3032 load_module: Unable to start Skype For Asterisk library. John Todd escreveu: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Aguirre (DagMoller) Infodag Consultoria FWD#: 459696 Enum#: +55 21 8871-4916 (e164.org) DUNDi-br#: 21 8871-4916 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nagios under *
Check the script permissions for nagios user
Sriram escreveu:
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to
show them line by line Up or down) using nagios using below script, but
i always get the status as down and red..can anyone let me know how to
read an output from nagios plugin ? nagios etc is configured already and
is working
PATH=/bin:/sbin:/usr/bin:/usr/sbin
FAILS=
STATUS=$(asterisk -rnx pri show span 1 | grep -a Status | awk
'{print $3;}' | cut -d, -f1)
if [ ${STATUS} == Up ]; then
echo PRI UP
exit 0
else
echo PRI DOWN
exit 2
fi
if i execute the script from command line i get the correct output i.e
OK for span 1 but on nagios web interface i get it as down...
If anyone can share the above script for asterisk monitoring then i wud
be grateful
rgds
Sriram
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Re: [asterisk-users] AMI - Status Event.
Simith, normaly, the caller channel have a minor uniqueid. Simith Nambiar escreveu: Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/192.168.1.100-091d7668 CallerID: 900 CallerIDNum: 900 CallerIDName: unknown Account: State: Up Link: SIP/192.168.1.100-b7400480 Uniqueid: 1225188315.137 ActionID: 65066874_3# Event: Status Privilege: Call Channel: SIP/192.168.1.100-b7400480 CallerID: 7 CallerIDNum: 7 CallerIDName: Vivas-Asterisk Account: State: Up *Context: amisim Extension: 900 Priority: 1 Seconds: 87* Link: SIP/192.168.1.100-091d7668 Uniqueid: 1225188315.136 ActionID: 65066874_3# Event: StatusComplete ActionID: 65066874_3# === I need to differentiate between the Caller and Callee so that i can display the Caller and Callee with different Images on a Graphical User interface. Looking at the Open Channels Status Events , i see the only difference between the 2 Status Events are the following fields: Context: amisim Extension: 900 Priority: 1 Seconds: 87 Is it safe to assume that the Status Event which holds the above fields are generally the Caller, i mean the person who dialed into the PBX ? Please let me know. Thank you. Cheers, Simith PS: Please redirect me to the right mailing list, if this list is inappropriate for such questions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Linux/Python 2.4.2] Forking Python doesn't work
Vincent,
try to use System() instead of AGI()
Diego Aguirre
Infodag - Informática
FWD#: 459696
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Vincent escreveu:
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so that users see the CID name + number pop up on
their computer screen, and simultaneously ring their phones.
The following script doesn't work as planned: It waits until the
script ends before moving on to the next step, which is Dial():
===
exten = s,1,AGI(netcid.py|${CALLERID(num)}|${CALLERID(name)}) exten
= s,n,Dial(${MYPHONE},5)
===
# cat netcid.py
#!/usr/bin/python
import socket,sys,time,os
def sendstuff(data):
s.sendto(data,(ipaddr,portnum))
return
sys.stdout = open(os.devnull, 'w')
if os.fork():
#BAD? sys.exit(0)
os._exit(0)
else:
now = time.localtime(time.time())
dateandtime = time.strftime(NaVm/%y NaVM, now)
myarray = []
myarray.append(STAT Rings: 1)
myarray.append(RING)
myarray.append(NAME + cidname)
myarray.append(TTSN Call from + cidname)
myarray.append(NMBR + cidnum)
myarray.append(TYPE K)
s = socket.socket(socket.AF_INET,socket.SOCK_DGRAM)
s.setsockopt(socket.SOL_SOCKET,socket.SO_BROADCAST,True)
portnum = 42685
ipaddr = 192.168.0.255
for i in myarray:
sendstuff(i)
#Must pause, and send IDLE for dialog box to close
time.sleep(5)
sendstuff(IDLE + dateandtime)
===
In another forum, people told me that I should fork twice. Is that
really necessary?
http://aspn.activestate.com/ASPN/Cookbook/Python/Recipe/278731
Thank you.
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[asterisk-users] Ignore This
Just checking... -- Diego Aguirre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Script AGI on C
Oi...
eu respondi sua mensagem na lista asteriskbrasil, mas com a moderação
dela, só deve chegar amanhã, hehehe
tenta um strip no arquivo.
# strip executable.agi
isso deve reduzir mais um pouco o tamanho do seu arquivo...
Diego Aguirre
Infodag - Informática
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Nikotel#: 99 21 8138-2710
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[EMAIL PROTECTED] escreveu:
Hi Folks:
I used that one example for AGI script on C web, only to fill the working with
the Asterisk. I compiled and it worked great. I executed accidentally the ls -l
command in directory where was the source and executable, I noted and was
surprised that because the executable size was to further 20 times more than
source.
I executed the gcc -Os source.c -o executable.agi command several times, with
otimization flags different. Maximum i can affort to reduce the executable size
was 17 times.
The source size full comment is 448 Bytes;
The size executable was about 7615 Bytes. (the maximum i got to reduce)
I was hope the executable size was in the order of magnitude of the proper
source size, since the comments are long.
Do one get to explain because of this?
Is this overhead consequence of linking with the operational system?
The script use only four functions of stdio.h library. It was seem that the
compiler include all stdio.h functions and compile all them.
I would like somebody of list to clear my doubt.
Regards,
Cleviton.
Here below small script used I grasp on site:
http://home.cogeco.ca/~camstuff/agi.html
/* C works just fine with Asterisk but you should use 'setlinebuf' on stdout and stderr. This causes buffering one line at a time
(rather than using a larger buffer). If you *don't* do this on stdout then your script will hang up while Asterisk waits for a
command but the (long) buffer isn't full yet. A minimal AGI script in C looks like this: */
//
#include stdio.h
main() {
charline[80];
/* use line buffering */
setlinebuf(stdout);
setlinebuf(stderr);
/* read and ignore AGI environment */
while (1) {
fgets(line,80,stdin);
if (strlen(line) = 1) break;
}
/* Send asterisk a command */
printf(SAY NUMBER 123 \\\n);
/* Read response from Asterisk and show on console */
fgets(line,80,stdin);
fputs(line,stderr);
}
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RES: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Hi Folks, Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. April, 1 - It´s a cool LIE!!! - Diego Aguirre FWD#: 459696 Tel/Enum: +55 21 2634-0968 -Mensagem original- De: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Em nome de Mike Hammett Enviada em: sexta-feira, 1 de abril de 2005 14:18 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now Remco Barende wrote: On Fri, 1 Apr 2005, Chris Hills wrote: Olle E. Johansson wrote: * New source code structure - C# and .net Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. Therefore, the code was quickly moved to C# on the .net platform. This gives Asterisk a lot of new features, including being fully integrated with Microsoft Exchange and Microsoft Active Directory. With all the user data stored in Active Directory, we finally have the user under full control. Users can dial in to the PBX to change their Windows password. We can also implement single-sign-on based on DTMF from a cell phone or WiFi phone. says Kelvin Reming. The C# language gives us much more modern code. And I'm so happy to get rid of the stupid-looking arctic bird, an ugly animal that that couldn't even fly. Shame this is just an april fool, I like the sound of this! Though it would be going head to head with Live Communications Server... I guess you missed the real joke there (the stability and secureness of .net) Ya, I mean do you really think an open source community is gonna acknowledge that MS can do anything right? of course not. THEY'RE THE DEVIL! (note, I will not respond to anything posted in reply to this, so don't even try) -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Budgetone 101 Hold/Xfer/Conf/Flash
Hold, transfer and flash only! the conference key is only for model 102-D Bill Michaelson escreveu: Is it possible to use the Hold/Transfer/Conference/Flash keys of the Budgetone-101 (FW 1.0.5.22) with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Aguirre FWD#: 459696 Tel/Enum: +55 21 2634-0968 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Best Grandstream firmware to use?
I'm using 1.0.5.18 with no problems. Diego Aguirre FWD# 459696 - Original Message - From: Yair Hakak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2005 6:55 AM Subject: Re: [Asterisk-Users] Best Grandstream firmware to use? i've actually had reboot issues since moving to 1.0.5.16, the phones seem to hang more often on soft reboot and require a hard reboot (unplugging). This is just a feeling and i can't quantify this but i don't remember having to physically reboot the phones this often before. I'm using one bt-101 and one bt-102. -yair On Tue, 18 Jan 2005 10:50:30 +0200, David Norton [EMAIL PROTECTED] wrote: I've been using 1.0.5.16 for more than a week now, haven't had a single problem, and have not had to reboot it once. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Tuesday, January 18, 2005 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best Grandstream firmware to use? I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11.It's relatively stable, and the last thing I want to do is update to a flaky firmware Paul -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Best Grandstream firmware to use?
No, i don't have this problem, the phone works fine. Diego Aguirre FWD# 459696 - Original Message - From: Leonardo Gomes Figueira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2005 10:22 AM Subject: Re: [Asterisk-Users] Best Grandstream firmware to use? Diego Aguirre wrote: I'm using 1.0.5.18 with no problems. 1.0.5.18 has an issue when registering (boot) and re-registering (after register expiration, 1 hour) that appears an 403 for a minute on the display (during this time the phone refuse calls) and then it comes back to normal operation. Didn't this happen with your phones ? Upgraded to 1.0.5.20 yesterday and I think this issue was fixed. Bye, Leonardo -- Leonardo Gomes Figueira [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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: Best Grandstream firmware to use?
http://fm.grandstream.com/gs/ Diego Aguirre FWD# 459696 - Original Message - From: Aldo Bergamini [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2005 12:21 PM Subject: [Asterisk-Users] Re: Best Grandstream firmware to use? [EMAIL PROTECTED] is believed to have said: Diego Aguirre wrote: I'm using 1.0.5.18 with no problems. 1.0.5.18 has an issue when registering (boot) and re-registering (after register expiration, 1 hour) that appears an 403 for a minute on the display (during this time the phone refuse calls) and then it comes back to normal operation. Didn't this happen with your phones ? Upgraded to 1.0.5.20 yesterday and I think this issue was fixed. Bye, Leonardo Leonardo, where did you get this firmware release? The Grandstream shows just 1.0.5.16 ... Thanks in advance Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Gotoif question
Try this:
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} == 800] ||
$[${CALLERIDNUM:0:3} = 866] || $[${CALLERIDNUM:0:3} = 877] ||
$[${CALLERIDNUM:0:3} = 888]?s|108)
Diego Aguirre
FWD# 459696
- Original Message -
From: John Hill [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 06, 2005 11:30 AM
Subject: [Asterisk-Users] Gotoif question
Is there a way to combine these lines into one?
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
exten = s,3,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
Thanks
--John
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Re: [Asterisk-Users] grandstream MWI?
in your sip.conf. voicemail=your extension you do not need to change grandstream configuration... Diego Aguirre - Original Message - From: Doug Reid - Stormcorp [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Tuesday, December 21, 2004 12:59 PM Subject: RE: [Asterisk-Users] grandstream MWI? HI How would I get the MWI working on the Grandsreams? Thanks Doug (Yip another one!) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Doug Lytle Sent: Monday, December 20, 2004 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream MWI? David Hajek wrote: Actually, I got the display flashing when I have a new message. But it is possible to get the Grandstream's Message button working? My goal is to pickup earphone and press Message button to retrieve my messages. David, I have both the message button and the MWI working under BETA 1.0.5.18 firmware. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] erroneous errors - registration fails forgrandstream phones
Hi, Look in your sip.conf host=192.168.20.2 and your phone is set to use 192.168.20.25 try to change host directive in sip.conf to host=192.168.20.25 Diego Aguirre - Original Message - From: Greg - Cirelle Enterprises [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 17, 2004 11:25 AM Subject: [Asterisk-Users] erroneous errors - registration fails forgrandstream phones Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to be erroneous since the phones/extensions apparently are working. example of 1 phone, but it happens with both: *** from asterisk CLI -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 Dec 17 08:01:59 NOTICE[22259]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' -- Saved useragent Grandstream BT100 1.0.5.20 for peer 40852 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 The date obviously changes *** from /var/log/asterisk/messages Dec 17 08:01:59 NOTICE[22259]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' The phones appear to work no traffic on the server 3Ghz P4 512MB RAM 75GB Free Disk Space Regards Greg Cirino ___ 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] tetting
testing ___ 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] BT-100 Transfer!!
greg wrote: Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel I use atendent transfer in Asterisk!!! Diego Aguirre Operações Internet - ramal 2563 Embratel - RJ - Original Message - From: Greg - Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 9:54 PM Subject: Re: [Asterisk-Users] BT-100 Transfer!! At 06:29 PM 12/9/04, you wrote: www.grandstream.com/BETATEST - Original Message - From: Mark Willis [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 11:36 PM Subject: RE: [Asterisk-Users] BT-100 Transfer!! I never could get attended transfer to work with the BT-100 on 1.0.5.16. Where did you get 1.0.5.18? It's not anywhere obvious on Grandstream's web site. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Thursday, December 09, 2004 02:56 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT-100 Transfer!! You need firmware 1.0.5.16 (Broken message button for voicemail) or 1.0.5.18 (Still in Beta, phone display '403' error about once per hour for 10 seconds or so. In order to use attended transfer you place the caller on hold by pressing the flash button and then dial the third person. Once you hang up the caller is transferred to the third person. Craig Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel greg ___ 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] BT-100 Transfer!!
I think that the conference only works for BT-102D. I have a BT-101. Diego Aguirre - Original Message - From: Mark Willis [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, December 10, 2004 1:16 PM Subject: RE: [Asterisk-Users] BT-100 Transfer!! That works for me too, just not obvious. Pity Conference doesn't work the same way. And 1.0.5.18 fixed the Message button, thanks to those who suggested the location. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Goddard Sent: Friday, December 10, 2004 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT-100 Transfer!! On Friday 10 December 2004 13:53, Greg - Cirelle Enterprises wrote: At 07:05 AM 12/10/04, you wrote: greg wrote: Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel Diego Aguirre wrote: I use atendent transfer in Asterisk!!! ok, the # extension key combo too bad the 3 buttons cannot be programmed to emulate the functions to make it work. i.e. the transfer button to send the key code of the # key, etc... same with conference and flash. It does work. Make sure on configuration page, you set send flash event to be NO, then pressing flash will not send any dtmf signal, but try to open another session. A talk to B, B press flash and hear dial tone, B dial C an talk to C. B press transfer to let A talk to C. then B hangs up. B ___ 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] BT-100 Transfer!!
No, i don't use the # key... 100 cal to 200 (BT-100), 200 press flash then 200 call to 300.. 200 talk to 300 and press transfer key (or hangup), now 100 talk to 300. the same is useful for Handytone ATA 286... sorry my english, this is not my language... Diego Aguirre Operações Internet - ramal 2563 Embratel - RJ - Original Message - From: Greg - Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, December 10, 2004 11:53 AM Subject: Re: [Asterisk-Users] BT-100 Transfer!! At 07:05 AM 12/10/04, you wrote: greg wrote: Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel Diego Aguirre wrote: I use atendent transfer in Asterisk!!! ok, the # extension key combo too bad the 3 buttons cannot be programmed to emulate the functions to make it work. i.e. the transfer button to send the key code of the # key, etc... same with conference and flash. Greg ___ 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] BT-100 Transfer!!
Hi, I have a BT-100 and Handytone ATA 286 with firmware 1.0.5.18. attended transfer works fine, message button broken has solved. Sorry my english, this is not my language. Diego Aguirre - Original Message - From: Mark Willis [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 1:36 PM Subject: RE: [Asterisk-Users] BT-100 Transfer!! I never could get attended transfer to work with the BT-100 on 1.0.5.16. Where did you get 1.0.5.18? It's not anywhere obvious on Grandstream's web site. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Thursday, December 09, 2004 02:56 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT-100 Transfer!! You need firmware 1.0.5.16 (Broken message button for voicemail) or 1.0.5.18 (Still in Beta, phone display '403' error about once per hour for 10 seconds or so. In order to use attended transfer you place the caller on hold by pressing the flash button and then dial the third person. Once you hang up the caller is transferred to the third person. Craig - Original Message - From: Altus Snyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 4:21 PM Subject: [Asterisk-Users] BT-100 Transfer!! Good day all We have Grand Stream BT-100 phones The transfer button work well, for blind transfer What the users want to do is, when a call comes in and asked to be transferred to another extension,for example 100,they 1ste want to speak to exten 100,then have the option transfer or not to transfer the call to this extension Currently they must pus flash for a new line speak to the pearson,flash again and then transfer? Any other or better ideas? Please Help Altus ___ 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] Restarting *
Title: Message Hi, To reload: # asterisk -r -x 'reload' to stop and start: # asterisk -r -x 'stop now' ; asterisk Diego Aguirre - Original Message - From: Ferguson, Michael To: [EMAIL PROTECTED] Sent: Thursday, December 02, 2004 12:50 PM Subject: [Asterisk-Users] Restarting * G'Day All What do I type at the command line to stop and start * on a RedHat ES3 box? Thanks ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Which modem is known to work with asterisk?
Hi, Intel Modems based on chipset Ambient MD3200 works fine! Diego Aguirre - Original Message - From: Michael Vogel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, November 24, 2004 6:46 AM Subject: [Asterisk-Users] Which modem is known to work with asterisk? Hi! I want to connect my asterisk system to the PSTN. Now I'm thinking about using an analog modem as FXO device. Which modems do run? How is the voice quality? (I looked into the wiki, but on http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware no regular modems were mentioned. Or should I use a special FXO-card? I don't want to spend 99$ for such a thing but I'm not sure if the X100P-clones for 30$ are working in germany. Thanks! 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
Re: [Asterisk-Users] Software SIP Phones
Hi, I am using X-Lite with Wine! Diego Aguirre - Original Message - From: Peter Osborne [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, November 17, 2004 2:47 PM Subject: [Asterisk-Users] Software SIP Phones Hi All, I'm curious to know what software based SIP phones people are using under Linux that work with Asterisk. I have tried several including kphone, linphone, and SJPhone, I have the same problem with all of them, my voice comes out quiet on the other end, and there is quite a bit of background noise making the call sound like a really bad cell phone. I would blame my onboard sound (I'm using a Toshiba M30 laptop) except that I have had no problems using Skype on this machine. Pete ___ 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
