Re: [Asterisk-Users] Re: Attended Transfer - transfer timeout, how to change?
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... you are using the attended transfer feature.. ist it already possible to hang up before the other person lifts the handset without loosing the caller when you are doing an attendet transfer? (person A takes an incoming call, person A would like to do an attended transfer to person B, person A hangs up the phone BEFORE person B takes the transfered call -- does the incoming call get lost?) this was an issue in 1.2.4, I'd like to know whether its fixed in 1.2.5. You shouldn't hang up. You should use disconnect = #0 from features.conf+ Yes - but thats not really comfortable :-( -- Tomislav Parcina tparcina#lama.hr ___ --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 ___ --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
[Asterisk-Users] asterisk and DDI
Hi, Somebody has someinfos forasterisk and swyx connected via DDI? Somebody has a example config for ddi wiith asterisk? regards rene ___ --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
[Asterisk-Users] Using IAX
Hello, I am a newbie, so I apologize for this maybe simple question. I want to connect two Asterisk machines with IAX. >From one machine I want to call to the other Asterisk,but sometimes I want to place the call on one context and sometimes in another one. I how can I do this?? When dialing on the dialplan? On the register= ... ? When defining the user in iax.conf??? I don't understand it well, sorry :( Thank you so much in advance, María ___ --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] g729 and latency measures
On Mon, 2006-03-20 at 11:38 +0530, ram wrote: Hi what is mtr ? where can i find that http://www.google.com/linux?hl=enlr=q=mtrbtnG=Search Pete ___ --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] Countries supporting SMS on PSTN (ISDN)
Unfortunately in Italy doesn't work: Italy and Spain uses Protocol Type2 and app_SMS doesn't support it (to my knowledge). http://www.rtx.dk/Files/Filer/tekniske%20artikler/SMStransmissionwithinthePS TN.pdf Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Robinson Sent: Saturday, 18 March 2006 02:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Countries supporting SMS on PSTN (ISDN) Hi does anyone have a definitive list of countries other than those listed on the wiki which are supporting app_SMS on landlines using ETSI ES 201 912 ?? ___ --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
[Asterisk-Users] Numbered Voicemails even with delete option!
Hello,
Thought people might be interested in this.
I want my voicemails emailed to a person and not stored on my asterisk
server. However, I want them to have a sequential number. I found that
if I set the option delete=1 in my voicemail.conf file for the mailbox,
then the numbering would keep being restarted.
I wrote this shell script to fool Asterisk into numbering my voicemails
sequentially even though I delete them. The files asterisk required are
replced with 2 byte files with the same name. It's these files that
asterisk uses for numbering the voicemails.
In voicemail.conf change your voicemail user so that emails are no
longer deleted.
1234 = 4242,Eicon Support,voicemail
Here is the shell script. Watch out for line wraps!! This script should
be run as a cron job at whatever interval you like.
NOTE: This is my first shell script so I'm sure it can be improved!
***
[EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean
cd /var/spool/asterisk/voicemail/default/1234/INBOX
#Only move files that are not currently in use that are over 3 bytes
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-size +3c -exec cp {}
/tmp \;
#replace contents of these files with 0 to save space
#find /tmp -name 'msg*.*' -and -type f -exec echo 0 {} \;
for i in /tmp/msg*.gsm
do
echo 0 $i
done
for i in /tmp/msg*.txt
do
echo 0 $i
done
if [ -f /tmp/msg\*.txt ]
then
rm -f msg\\*.txt
rm -f msg\\*.gsm
fi
#delete any wav or WAV files
rm -f /tmp/*.wav
rm -f /tmp/*.WAV
#Delete any files that are not currently being used
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-type f -and -size +3
c -exec rm -f {} \;
#Copy our changed files back to the directory to fool asterisk!
cp -f /tmp/msg*.* /var/spool/asterisk/voicemail/default/1234/INBOX/
rm -rf /tmp/msg*.*
#Cleanup
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.gsm
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.txt
***
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[Asterisk-Users] Grabbing the billsec and duration after a hangup.
Hello,
I am wondering if someone has got any ideas that can help solve this
problem.
I have a dial plan that you call into, and depending on certain conditions
it calls out on a number grabbed from a database.
Something like this :
exten = s,n,Do something
exten = s,n,Do something else
exten = s,n,Dial(ZAP/g1/${OUTBOUND},${timeout})
I need to log the time the person was connected to $(OUTBOUND) , these are
duration and billsec in the CDR's
So at hangup I do something like this.
exten = h,1,DeadAGI(cdr-
outlogger.php|${CDR(start)}|${OUTBOUND}|${CDR(channel)}|${CDR(duration)}|$
{CDR(billsec)}|${CDR(disposition)}|${CDR(accountcode)})
Trouble is duration and billsec are *ALWAYS* 0 (zero), as if they have not
been loaded with the values, even though the channel is hung up.
Anyone got any ideas on how I can access ${CDR(duration)} and
${CDR(billsec)} in the hangup extension?
Thanks, hope I explained that well enough.
Mark
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[Asterisk-Users] Problems loading res_odbc.so and cdr_odbc.so
Hi. I am having troubles loading the res_ and cdr_odbc modules, they fail because they cannot find libodbc.so.1 I have unixODBC properly installed and the needed DNS setup correctly. Any ideas why I am having this troubles? Where is asterisk looking for the libodbc.so.1 file? And were can I configure this path? Thanks in advance. ___ --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] Numbered Voicemails even with delete option!
On Mon, 2006-03-20 at 09:32 +, David Waugh wrote:
NOTE: This is my first shell script so I'm sure it can be improved!
noted, in that spirit see notes below ...
***
[EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean
cd /var/spool/asterisk/voicemail/default/1234/INBOX
this appears to be redundant since you specify the full path in the find
below ... It doesnt hurt anything though.
#Only move files that are not currently in use that are over 3 bytes
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-size +3c -exec cp {}
/tmp \;
why cp em to /tmp? Seems a waste given what you do with the files
in /tmp later...
#replace contents of these files with 0 to save space
#find /tmp -name 'msg*.*' -and -type f -exec echo 0 {} \;
for i in /tmp/msg*.gsm
do
echo 0 $i
done
for i in /tmp/msg*.txt
do
echo 0 $i
done
if [ -f /tmp/msg\*.txt ]
then
rm -f msg\\*.txt
rm -f msg\\*.gsm
fi
after putting a 0\n in each file you then rm it without doing anything
else ... Why waste the time copying them earlier, then making the files
contain only 0\n just to rm em?
#delete any wav or WAV files
rm -f /tmp/*.wav
rm -f /tmp/*.WAV
#Delete any files that are not currently being used
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-type f -and -size +3
c -exec rm -f {} \;
#Copy our changed files back to the directory to fool asterisk!
cp -f /tmp/msg*.* /var/spool/asterisk/voicemail/default/1234/INBOX/
rm -rf /tmp/msg*.*
But if those exited they were deleted above ...
#Cleanup
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.gsm
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.txt
If by some chance they were able to survive the previous copies,
deletes, then copied again, you make sure they dont survive any
further :)
how about this, would it do what you want (note I am basically using
what you started out with) It also makes em 0 bytes instead of 2 :)
And it works on more than one user at a time, although that may not be
desired.
The size +3c may need to be altered since it wont have 2 bytes, but it
shouldnt hurt anything to leave it, I left it becuase that is what you
started out doing, although for other reasons.
touch /tmp/vm.$$
for i in /var/spool/asterisk/voicemail/default/*; do
find $i/INBOX -mmin +1 -and -size +3c -and -name \*.wav \
-exec rm {} \;
find $i/INBOX -mmin +1 -and -size +3c -and -name \*.WAV \
-exec rm {} \;
find $i/INBOX -mmin +1 -and -size +3c \
-exec cp -f /tmp/vm.$$ {} \;
done
rm /tmp/vm.$$
This of course could still be optimized further, but to keep it simple I
decided to use what you originally did as a base...
app_voicemail can detect a gap in the sequencing between any rm and
creation of a replacement file, so I create a dummy file in /tmp then cp
that over the desired file to avoid that.
find is not that processor friendly so you will want to watch out if you
have a large number of users/voicemails.
I also dont see that big of a point in doing this, after how many years
and hundreds of thousands of voicemails that a user has listened to do
you finally reset that number? or do you want a life count forever?
Further the way app_voicemail works the larger that directory is the
more processing that is required to find the next available sequence
number ...
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
signature.asc
Description: This is a digitally signed message part
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[Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h -- OctoBRI -- PABX e-Generis ISDN Phones | | SIP Phones France Telecom -- SIP Phones : Works France Telecom -- ISDN Phones : Works SIP Phones -- ISDN Phones : Works ISDN Phones - SIP Phones : Works SIP Phones -- France Telecom : DOESN'T WORK ISDN Phones - France Telecom : DOESN'T WORK Here are some characteristics of my Asterisk Setup OS Linux Gentoo 2.6.15-r1 zaptel 1.2.3 libpri 1.2.2 asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h ISDN Lines : EuroISDN not EuroISDN+ Junghanns OctoBRI PCI ISDN Card S/T 1+8 - S/T 2+7 : TE Mode S/T 3+6 - S/T 4+5 : NT Mode modprobe qozap ports=60 zaptel.conf --- loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 --- zapata.conf --- switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres = yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callprogress=yes context=isdn-incoming group = 1 ; S/T port 1,2,7,8 channel = 1-2 channel = 4-5 ;channel = 19-20 channel = 22-23 context=pbx-incoming group = 2 channel = 7-8 channel = 10-11 ;channel = 13-14 channel = 16-17 - Here's the output BRI debug when I try to make outbound calls from a SIP phone : -- Executing Dial(SIP/400-c8dc, Zap/1/1013) 1 -- Making new call for cr 137 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (Cool len=26 1 Call Ref: len= 1 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 811 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 051 411 801 341 301 301 ] 1 Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1 Presentation: Presentation permitted, user number not screened (0) '400' ] 1 [1 701 051 c11 311 301 311 331 ] 1 Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] -- Called 1/1013 1 Protocol Discriminator: Q.931 (Cool len=8 1 Call Ref: len= 1 (reference 137/0x89) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 871 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) 1 Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] 1 -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup, cause 100 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/400-c8dc, ) == Spawn extension (default, 1013, 2) exited non-zero on 'SIP/400-c8dc' 1 received TEI check request for TEI = 127 I've already tested several configurations for zapata.conf especially with the pridialplan and switchtype lines but without success. Could you help me to analyse and solve this odd problem ? Thank you in advance, -- Sébastien Mortier AbsysTech Tel : +33 3 20 50 99 02 Fax : +33 3 20 74 50 05 Gsm : +33 6 20 79 24 29 http://www.absystech.fr ___ --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
[Asterisk-Users] simple perl-agi - where's the error?
Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian ___ --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
[Asterisk-Users] Help: Using asterisk and mysql for a university project
Hello all, I want to use mysql for to save the users of my asterisk PBX. I use the realtime solution with mysql but when I made the sip show peers command doesnt appear my users. My configurations are: res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = satec dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock sip.conf [general] dbuser=asterisk dbpass=satec dbhost=127.0.0.1 dbname=asterisk table=sipusers rtcachefriends=yes extconfig.conf sipusers = mysql, asterisk, sipusers sippeers = mysql, asterisk, sipusers and the output of the realtime mysql status asterisk2006*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username asterisk for 3 minutes, 42 seconds. asterisk2006*CLI sip show peers Name/username Host Dyn Nat ACL Port Status 0 sip peers [0 online , 0 offline] asterisk2006*CLI sip show users Username Secret Accountcode Def.Context ACL NAT asterisk2006*CLI What is the problem? Thanks and Regards, Sergio ___ --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
[Asterisk-Users] Re: Local Channel
In article [EMAIL PROTECTED], Darren Wiebe [EMAIL PROTECTED] wrote: I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Make sure you are specifying it as /n, and not \n like you wrote above. If it still doesn't work, then it is most likely a bug... Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Numbered Voicemails even with delete option!
Thanks Bret for the input. Your solution seems a lot neater=)
I had problems with globbing I think it is called.
I kept getting files name being created called msg*.txt which caused
me problems later.
I think your way removes this.
The reason I was doing this was for testing purposes. I was making many
thousands of calls to the asterisk server as a stress test, and then
emailing them across. - Nothing really useful in the real world.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: 20 March 2006 10:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Numbered Voicemails even with delete
option!
On Mon, 2006-03-20 at 09:32 +, David Waugh wrote:
NOTE: This is my first shell script so I'm sure it can be improved!
noted, in that spirit see notes below ...
***
[EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean
cd /var/spool/asterisk/voicemail/default/1234/INBOX
this appears to be redundant since you specify the full path in the find
below ... It doesnt hurt anything though.
#Only move files that are not currently in use that are over 3 bytes
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-size +3c -exec cp {}
/tmp \;
why cp em to /tmp? Seems a waste given what you do with the files
in /tmp later...
#replace contents of these files with 0 to save space
#find /tmp -name 'msg*.*' -and -type f -exec echo 0 {} \;
for i in /tmp/msg*.gsm
do
echo 0 $i
done
for i in /tmp/msg*.txt
do
echo 0 $i
done
if [ -f /tmp/msg\*.txt ]
then
rm -f msg\\*.txt
rm -f msg\\*.gsm
fi
after putting a 0\n in each file you then rm it without doing anything
else ... Why waste the time copying them earlier, then making the files
contain only 0\n just to rm em?
#delete any wav or WAV files
rm -f /tmp/*.wav
rm -f /tmp/*.WAV
#Delete any files that are not currently being used
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-type f -and -size +3
c -exec rm -f {} \;
#Copy our changed files back to the directory to fool asterisk!
cp -f /tmp/msg*.* /var/spool/asterisk/voicemail/default/1234/INBOX/
rm -rf /tmp/msg*.*
But if those exited they were deleted above ...
#Cleanup
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.gsm
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.txt
If by some chance they were able to survive the previous copies,
deletes, then copied again, you make sure they dont survive any
further :)
how about this, would it do what you want (note I am basically using
what you started out with) It also makes em 0 bytes instead of 2 :)
And it works on more than one user at a time, although that may not be
desired.
The size +3c may need to be altered since it wont have 2 bytes, but it
shouldnt hurt anything to leave it, I left it becuase that is what you
started out doing, although for other reasons.
touch /tmp/vm.$$
for i in /var/spool/asterisk/voicemail/default/*; do
find $i/INBOX -mmin +1 -and -size +3c -and -name \*.wav \
-exec rm {} \;
find $i/INBOX -mmin +1 -and -size +3c -and -name \*.WAV \
-exec rm {} \;
find $i/INBOX -mmin +1 -and -size +3c \
-exec cp -f /tmp/vm.$$ {} \;
done
rm /tmp/vm.$$
This of course could still be optimized further, but to keep it simple I
decided to use what you originally did as a base...
app_voicemail can detect a gap in the sequencing between any rm and
creation of a replacement file, so I create a dummy file in /tmp then cp
that over the desired file to avoid that.
find is not that processor friendly so you will want to watch out if you
have a large number of users/voicemails.
I also dont see that big of a point in doing this, after how many years
and hundreds of thousands of voicemails that a user has listened to do
you finally reset that number? or do you want a life count forever?
Further the way app_voicemail works the larger that directory is the
more processing that is required to find the next available sequence
number ...
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] Call Pickup Woes
What about setting up DYNAMIC_FEATURES=pickupexten inside your [globals] ? This is needed for, as the variable name says, dynamic features. And don't forget to set callgroup/pickupgroup to each one in your sip.conf Does anyone tested the new application Pickup()? []'s MM On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I’ve done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed? ___ --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 ___ --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
[Asterisk-Users] MixMonitor and transferred calls
Hi; I'm trying to record all inbound and outbound calls at a site, and I have a problem with inbound calls that are transferred by a receptionist using Snom's handset buttons (i.e. SIP transfer rather than using the key sequences defined in features.conf). The first leg of the call is recorded fine. There is, however, no recording after the transfer. Am I correct in thinking that I'll have to use Asterisk native transfer for this to work ? jd ___ --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] simple perl-agi - where's the error?
Try setting it to sth like SIP/200 instead of a single number. l. On Mon, 20 Mar 2006 11:56:50 +0100, Christian B [EMAIL PROTECTED] wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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
AW: [Asterisk-Users] simple perl-agi - where's the error?
Tried: $DIALSTRING??? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Lenz Gesendet: Montag, 20. März 2006 12:56 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] simple perl-agi - where's the error? Try setting it to sth like SIP/200 instead of a single number. l. On Mon, 20 Mar 2006 11:56:50 +0100, Christian B [EMAIL PROTECTED] wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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 ___ --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
[Asterisk-Users] AMP and ABE
Hi, Has anyone had experience installing AMP/FreePBX on Asterisk Business Edition? The main issue we have come across is FreePBX requires a dependency PHP-PEAR PHP-GD which is not available on RedHat RHEL3 (ES) Thanks James ___ --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] Annoying Asterisk Realtime Limitation
snip Anyway, so I went back to a plain text file for sip.conf. What a dissapointment. /snip This is kind of backwards but you can make a script that will pull all the info from the DB and save it as sip.conf. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] simple perl-agi - where's the error?
of course, but this doesn't make the difference(i just simplified the input-variable to verify it's not a regexp-issue). It should at least try to use to dial the single number i've set, but it looks like the variable is empty... On Mon, 20 Mar 2006 12:55:38 +0100 Lenz [EMAIL PROTECTED] wrote: Try setting it to sth like SIP/200 instead of a single number. l. On Mon, 20 Mar 2006 11:56:50 +0100, Christian B [EMAIL PROTECTED] wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian ___ --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
[Asterisk-Users] How to setup Proxy info to * box , [* box behind a squid proxy and firewall ]
Hi All I had successfully tried out asterisk on the LAN , now I want to call outside using sipdiscount or using http://exgn.net my asterisk box is behing a Firewall and the Internet usage is through a proxy server located at 192.168.20.20:8080 Now I want to configure asterisk box in this setup ( proxt and then firewall ) , the port 4569 ,5060,5036 are opened by firewall and the proxy Now I need advice , 1 On which file to edit , to tell asteriskbox to use my proxy 192.168.20.20:8080 server to connect outside 2 If I can give the proxy server details to asteriskbox , will I be able to dial to the outside world Thanks Joseph John ___ Yahoo! Photos NEW, now offering a quality print service from just 8p a photo http://uk.photos.yahoo.com ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ --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] simple perl-agi - where's the error?
no, this doesn't make a difference On Mon, 20 Mar 2006 13:01:00 +0100 René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote: Tried: $DIALSTRING??? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Lenz Gesendet: Montag, 20. März 2006 12:56 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] simple perl-agi - where's the error? Try setting it to sth like SIP/200 instead of a single number. l. On Mon, 20 Mar 2006 11:56:50 +0100, Christian B [EMAIL PROTECTED] wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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 ___ --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 ___ --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] Call Pickup Woes
Melcon Moraes wrote: On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I’ve done this, however there is no definition for *8 in extensions.conf. I've confirmed this morning. Call pickup is broken in 1.24. I've upgraded our system to 1.25 over the weekend and tested out call pickup this morning. It now works. Doug ___ --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
[Asterisk-Users] How to make caller groups ???
Hello All !!!I have 4 PSTNlines in the PBX server 1,2,3,4. Firstline will be usedby only one extension (i.e. for the boss) for incom ing and outgoing. This line is dedicated for him only.(The remaining lines will be shared bythe employees 1) Group Ahave access to lines 2 , 3 4. 2)Group Bhave access tolines 3 4 3)Group C have access to line 4I want to know that how i will make that groups. I will be grateful for ur help.Thanks a lot. Faisal Relax. Yahoo! Mail virus scanning helps detect nasty viruses!___ --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
[Asterisk-Users] answer delay
Hi guys, maybe youìve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller-telco(telco bill to the caller as soon as * answer)-asterisk Thanks in advance. -- .:FaberK:. ___ --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] Asterisk Users Mailing List Traffic
One thing that may help: I use outlook rule to move all the messages into a folder. Then outlook has a feature, instead of sorting by date, or subject, you can sort by conservation. It then groups the messages by thread in date order, so you can sort through the emails very quickly and allows you to switch digest off. Thanks James ___ --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] Call Pickup Woes
And don't forget to set callgroup/pickupgroup to each one in your sip.conf Call pickup works among IAX phones? Mimmus ___ --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] How to enable talking in chanspy while spying?
I also may be able to contribute as well. Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Sunday, 19 March 2006 5:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to enable talking in chanspy while spying? Does anyone know how much was paid ? We would be willing to part-fund this and to release it as part of the distribution. Julian. Steven Totaro wrote: This is an age old question. Unless something has changed, it is possible but not included functionality. A group of people paid to have this functionality developed but since they paid they decided not to release it back into the asterisk community. I am not sure if it for sale or not or even if it is, what the cost is. If you are listening to a zap channel with zapscan, it works like we want but not with chanspy (my understanding anyways). I need the same functionality for my call center so if you find a solution (even if it has to be purchased) please post back to the list. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of atik khan Sent: Saturday, March 18, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to enable talking in chanspy while spying? hello i want to spy on a chennel listen the voice conversation between two person. i also want talk to one of them but others will not listen my voice. how can i configure this using ChanSpy? thanks atik ___ --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 ___ --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 ___ --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 ___ --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: RE : [Asterisk-Users] TDM 2400 With 24 FXO
Dear Francois, Thanks for your advise,, I'll buy the echocan module Best Regards, Fernando -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 18, 2006 6:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] TDM 2400 With 24 FXO Hello Fernando, I have checked this card with and without hardware echocan : the hardware echocan module does the job better than the zaptel software can do it. I recommand this module without any doubt. But, the echocan algorithms in zaptel are better and better and the CPUs power grows permanently. It is possible to use this card without hardware echocan, but you will encounter the same results, in this case, as you can obtain with the other TDM Digium's cards : correct for certain situations, not for all extreme cases, depending what listening level your users want, lines specifications and what critical echo threshold they can admit before to not be able to do correctly their job. Near same thing for E1/T1 harware echocan features. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Fernando BERRETTA Envoyé : vendredi 17 mars 2006 14:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] TDM 2400 With 24 FXO Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando ___ --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 ___ --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 ___ --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] AMP and ABE
James Sturges wrote: Hi, Has anyone had experience installing AMP/FreePBX on Asterisk Business Edition? The main issue we have come across is FreePBX requires a dependency PHP-PEAR PHP-GD which is not available on RedHat RHEL3 (ES) Thanks James ___ --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 google is your friend http://us3.php.net/gd http://pear.php.net/ -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . signature.asc Description: OpenPGP digital signature ___ --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] Zaptel will not build
Hello I've been trying to compile zaptel 1.2.4 on Mandriva 10.2 , kernel 2.6.11-6mdk and i keep getting these errors: #make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file ZAPTELVERSION=1.2.4 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.11-6mdk/build make -C /lib/modules/2.6.11-6mdk/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/lib/modules/2.6.11-6mdk/build' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/lib/modules/2.6.11-6mdk/build' make: *** [linux26] Error 2 I've made all the modifications to the spinlock.h file [ #define DEFINE_RWLOCK(x) rwlock_t x = RW_LOCK_UNLOCKED] suggested on this list and also the changes to the Zaptel Makefile [CFLAGS+=$(shell if uname -r | grep -q 2.6.11-6mdk ; then echo -Drw_lock_t=\rwlock_t\; fi) ] , but the problem persists . Any ideas ? -- Assaf Flatto Atelis IT Manager Cellular: +972-54-5679230 e-mail: [EMAIL PROTECTED] ___ --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] A general deployment question (OT)
There are about 10k subscribers to this list, there is a good number to start with. On 3/18/06, Rob Gillan [EMAIL PROTECTED] wrote: Does anyone have a guesstimate of how many active Asterisk installations there are? Sorry this is off topic, need it for a customer proposal and they need comfort on stability. A count of the downloads from Digium would be a good start but I couldn't find this anywhere with Google. Feedback from anyone who may know near actual data would be appreciated rather than simply guessing, as I hope this doesn't generate too many posts (sorry if it does). Cheers Rob ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] An FXO version of IAXy?
Also note the use of FXO for Overhead Paging needs. Most all systems from Valcom, Bogen and some others are C.O. Line only and the line converter can cause huge delays in broadcast.. On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote: In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? There's a hugh market for such a box, and none of the current manufacturers have addressed the one-to-four pstn line boxes with anything that would be considered reasonable quality. The GS 488 appears to be their 'test-the-market' box, but its not very usable based on my testing. The Mediatrix 1204 does an excellent job with audio, but is over-priced and under-supported from my perspective. The spa3k comes the closest to providing a reasonable interface with acceptable audio, but has several functions that really need to be fixed. I hope the Linksys folks address those issues instead of dropping the box. From my perspective, designing a fxo box that can interface to the many country standards and has a reasonable echo canceller is not an easy task. Much more difficult than designing a fxs box. And, if you look at the cost of the hardware echo canceller chips that can support 128 taps, the manufacturing cost of a fxo box becomes rather expensive. If you look at the market from a manufacturer's perspective, the sales of fxo boxes are significantly less then the sales of fxs boxes. Therefore it makes sense what the majority of them are doing from an RD and manufacturing perspective (eg, address the larger market before incurring the expense of the smaller market). ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] AMP and ABE
Google is a good friend, unfortunately the system admin who represent the company we are installing is not so. They a requiring an audited stable platform, aka Asterisk Business Edition. So when we say we need to install non-certified package onto their Enterprise Server, they say na! Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Monday, 20 March 2006 11:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP and ABE James Sturges wrote: Hi, Has anyone had experience installing AMP/FreePBX on Asterisk Business Edition? The main issue we have come across is FreePBX requires a dependency PHP-PEAR PHP-GD which is not available on RedHat RHEL3 (ES) Thanks James ___ --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 google is your friend http://us3.php.net/gd http://pear.php.net/ -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . ___ --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] simple perl-agi - where's the error?
You should try '$res = $AGI-exec(DIAL, $dialstring);' Christian B wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Benoit Merouze Network Software Developer [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP and ABE
On 20/03/06, James Sturges [EMAIL PROTECTED] wrote: Google is a good friend, unfortunately the system admin who represent the company we are installing is not so. They a requiring an audited stable platform, aka Asterisk Business Edition. So when we say we need to install non-certified package onto their Enterprise Server, they say na! Then shouldn't you be requesting support from the supplier of that audited, stable platform, instead of requesting community support? Isn't that why you (they) bought it? No much point otherwise. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pickup problem
Hello,I can pickup a call from a specific number:exten = _8XXX, 1, Pickup(${EXTEN:1})But i couldnt pickup calls coming from PSTN to local extensions.Another question is it possible to pickup the last calling number without any exten.Can you help me?erkaN
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Re: [Asterisk-Users] A general deployment question (OT)
http://www.asterisk.org/node/36 Boasting close to a quarter-million users in over 200 countries... MATT--- On 3/20/06, Andrew Latham [EMAIL PROTECTED] wrote: There are about 10k subscribers to this list, there is a good number to start with. On 3/18/06, Rob Gillan [EMAIL PROTECTED] wrote: Does anyone have a guesstimate of how many active Asterisk installations there are? Sorry this is off topic, need it for a customer proposal and they need comfort on stability. A count of the downloads from Digium would be a good start but I couldn't find this anywhere with Google. Feedback from anyone who may know near actual data would be appreciated rather than simply guessing, as I hope this doesn't generate too many posts (sorry if it does). Cheers Rob ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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 ___ --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] Zaptel will not build
On Mon, Mar 20, 2006 at 03:38:21PM +0200, Assaf Flatto wrote: Hello I've been trying to compile zaptel 1.2.4 on Mandriva 10.2 , kernel 2.6.11-6mdk and i keep getting these errors: #make linux26 [ snip ] /lib/modules/2.6.11-6mdk/build make -C /lib/modules/2.6.11-6mdk/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/lib/modules/2.6.11-6mdk/build' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/lib/modules/2.6.11-6mdk/build' make: *** [linux26] Error 2 Have you edited the makefile in any way? The error is because MODULES is set to an empty value. -- Tzafrir ___ --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] Asterisk Users Mailing List Traffic
On 3/19/06, James Harper [EMAIL PROTECTED] wrote: That being said, a mailing list with a forum interface (or a forum witha mailing list option) might be a reasonable compromise as it should meet the needs of both mailing list lovers and forum lovers (assuming itis implemented properly!) All- if you haven't tried gmail for mailing lists you are missing the boat. I have over two years of this mailing list in my gmail account now and I can't imaging not having that resource at my fingertips. vi-like keyboard navigation 2+gig storage browser accessible (including wap) threading google search capabilities rule based archiving etc etc etc -Brian ___ --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] Asterisk Users Mailing List Traffic
On Sat, Mar 18, 2006 at 08:23:03PM -0600, Rich Adamson wrote: This same issue has been discussed many times over the last two years. Not likely its going to change now. I just love this attitude. Could someone managing these lists outline the requirements to change the lists? Do we need a vote or is it something which can be simply adopted once the right people recognize the problem (and if so, who is the right people)? Aaron Daniel wrote: Splitting the list by type of request may be a good idea, but splitting based on skill level is just a bad idea... I'm pretty sure that regardless of a newbie's status, they'll still just go to the other lists as the newbie list likely won't do much good. In short, I agree with different lists for hardware and configuration questions... Aaron On Mar 18, 2006, at 4:05 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --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 ___ --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 ___ --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 ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] How to enable talking in chanspy while spying?
What about the Monitor command from the manage api. It allows for monitoring but not coaching. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Totaro Sent: Saturday, March 18, 2006 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How to enable talking in chanspy while spying? This is an age old question. Unless something has changed, it is possible but not included functionality. A group of people paid to have this functionality developed but since they paid they decided not to release it back into the asterisk community. I am not sure if it for sale or not or even if it is, what the cost is. If you are listening to a zap channel with zapscan, it works like we want but not with chanspy (my understanding anyways). I need the same functionality for my call center so if you find a solution (even if it has to be purchased) please post back to the list. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of atik khan Sent: Saturday, March 18, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to enable talking in chanspy while spying? hello i want to spy on a chennel listen the voice conversation between two person. i also want talk to one of them but others will not listen my voice. how can i configure this using ChanSpy? thanks atik ___ --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 ___ --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 ___ --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] Asterisk Users Mailing List Traffic
Kristian Larsson wrote: On Sat, Mar 18, 2006 at 08:23:03PM -0600, Rich Adamson wrote: This same issue has been discussed many times over the last two years. Not likely its going to change now. I just love this attitude. Guess its not an attitude as much as having been around this list for over two years, and listening to the same old topic coming up every six months without looking back at the previous discussions on the topic. Could someone managing these lists outline the requirements to change the lists? Do we need a vote or is it something which can be simply adopted once the right people recognize the problem (and if so, who is the right people)? You might try Kevin at digium. The age-old argument/discussion relative to the -user list is that if you split it into pieces (eg, newbies plus other lists), no one will hang around the newbie list to answer questions, and the newbies will migrate to other lists. (Just about like newbies posting to the -dev list when they don't get a quick answer on the -user list, or, the @home folks posting to the -user list when few of the -user list members actually use @home.) FWIW, I'd vote to keep it the way it is now and I'll just make use of the delete key to handle uninteresting noise. ___ --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] 7970 Configs
Hi,
I just download the SIP image (cmterm-7970_7971-sip.8-0-2-0.cop) from
Cisco, copy all files on my tftpboot, create a SEP{mac}.cnf.xml file
(take the one posted by Greg Oliver) with some modification.
If the secret= is empty on the server, I receive now request on the
Asterisk server but the phone send the request 2-3 times per second to
the server.
(Repeated request...)
-- Registered SIP '1009' at xx.xx.xx.247 port 49504 expires 3600
-- Registered SIP '1009' at xx.xx.xx.247 port 49505 expires 3600
-- Registered SIP '1009' at xx.xx.xx.247 port 49506 expires 3600
(.)
(Register Request)
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.136:5060;branch=z9hG4bK4dc6894b
From: sip:[EMAIL PROTECTED];tag=0015f97f42710003b4619858-cc0ebb79
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Max-Forwards: 70
Date: Mon, 27 Feb 2006 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact:
sip:[EMAIL PROTECTED]:5060;transport=udp;+sip.instance=urn:uuid:----0015f97f4271;+u.sip!model.ccm.cisco.com=30006
Since my phone is behind a NAT, I have enable these setting:
natReceivedProcessingtrue/natReceivedProcessing
natEnabledtrue/natEnabled
natAddress/natAddress
I can establish an SSH connection to the phone (sshUserID/sshPassword)
and can log into phone (debug/debug and log/log) to get more
informations (like show config)
For SIP Proxy Authentication, with show config, I see a setting for
authPassword, but try to put it on the xml file but doesnt work.
I have never used CallManager, I presume that we need this to generate a
template SEP cnf.xml files ? Not found any documentations on Cisco
site about SIP or SCCP parameters that must be in this file.
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
On Monday 20 March 2006 09:39, Kristian Larsson wrote: I just love this attitude. A modicum of thought for others may save you from yourself. This has been discussed many, many times. The problem is a complex one, and one that has been thought through many times by people much smarter than you and I. The general consensus has been that having 50 lists is *no* better than having the three (more if you count the svn commits, documentation, etc.) we have now. Why can you not understand this, or at least read over the archives which point this out and contain the entire discussion regarding the problem? Why do you instead post this snarky, sarcastic comment? Do we need to cater to you and rehash all the discussion just for your benefit? Could someone managing these lists outline the requirements to change the lists? Read the archives. Do your own homework. Do we need a vote or is it something which can be simply adopted once the right people recognize the problem (and if so, who is the right people)? Again, read the archives. The answers and reasons behind this are all there. -A. ___ --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] pickup problem
erkan kolemen wrote:
Hello,
I can pickup a call from a specific number:
exten = _8XXX, 1, Pickup(${EXTEN:1})
But i couldnt pickup calls coming from PSTN to local extensions.
I'm using a dialplan entry like yours:
exten = _*9,1,Pickup(${EXTEN:2})
and just tested it. Working fine using svn trunk as of yesterday.
Another question is it possible to pickup the last calling number
without any exten.
Not sure what you're asking. Your example above is directed call
pickup, but there is also a more generic call pickup using
'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach
uses *8 or *8# to pickup any ringing phone within the callgroup number
(eg, 2 in this example).
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Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
On Monday 20 March 2006 09:56, Rich Adamson wrote: FWIW, I'd vote to keep it the way it is now and I'll just make use of the delete key to handle uninteresting noise. Amen. I currently have 12619 messages in -users, and that's with kmail expiring old messages. I've been on these lists as long as you and feel that this is simply the best way there is. I've been involved in projects with eighty mailing lists to cover every facet of the project to try and keep traffic down and it Simply Does Not Work. Anyone claiming otherwise has never been involved in such a project. -A. ___ --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] Re: problems with emailing voicemail
Guys, Thanks again for all your help. I've updated /etc/sysconfig/network and /etc/hosts as per your suggestions: /etc/sysconfig/network: NETWORKING=yes HOSTNAME=localhost.localdomain 127.0.0.1 my external, static IP address asterisk localhost /etc/hosts: # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 localhost.localdomain localhost 127.0.0.1 my external, static IP address asterisk localhost But still no luck. I've also noticed a problem when I boot the server. Sendmail along with a number of other processes generate an errror message. They're all pretty much the same error message and same line number so here's an example: etc/rc.d/rc.sysinit error line 3: 127.0.0.1 command not found FWIW, I've included the first part of my etc/rc.d/rc.sysinit below. I'm stumped. This was working before, what could have possibly happened? Any other ideas on how to fix it before I go and re-install RedHat?? Thanks, Hugh #!/bin/bash # # /etc/rc.d/rc.sysinit - run once at boot time # # Taken in part from Miquel van Smoorenburg's bcheckrc. # # Rerun ourselves through initlog if [ -z $IN_INITLOG -a -x /sbin/initlog ]; then exec /sbin/initlog $INITLOG_ARGS -r /etc/rc.d/rc.sysinit fi # If we're using devfs, start devfsd now - we need the old device names [ -e /dev/.devfsd -a -x /sbin/devfsd ] /sbin/devfsd /dev HOSTNAME=`/bin/hostname` if [ -f /etc/sysconfig/network ]; then . /etc/sysconfig/network else NETWORKING=no fi if [ -z $HOSTNAME -o $HOSTNAME = (none) ]; then HOSTNAME=localhost fi . /etc/init.d/functions # Start the graphical boot, if necessary if [ $BOOTUP = graphical ]; then if [ -x /usr/bin/rhgb ]; then /usr/bin/rhgb else export BOOTUP=color fi fi last=0 for i in `LC_ALL=C grep '^[0-9]*.*respawn:/sbin/mingetty' /etc/inittab | sed 's/^.* tty\([0-9][0-9]*\).*/\1/g'`; do /dev/tty$i last=$i done if [ $last -gt 0 ]; then /dev/tty$((last+1)) /dev/tty$((last+2)) fi if [ `/sbin/consoletype` = vt -a -x /sbin/setsysfont ]; then echo -n Setting default font ($SYSFONT): /sbin/setsysfont if [ $? -eq 0 ]; then success else failure fi echo ; echo fi On 3/17/06, Colin Anderson [EMAIL PROTECTED] wrote: if you are using Sendmail, then you have to add a trusted user to /etc/sendmail.cf in the format: Tuser So if the user you run Asterisk under is called asterisk then you add Tasterisk to sendmail.cf. Restart Sendmail. Then in voicemail.conf you can set the email address to whatever you want, it's at the top of the config file, I believe. -Original Message-From: hugolivude [mailto:[EMAIL PROTECTED]] Sent: Friday, March 17, 2006 10:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: problems with emailing voicemailWow, Thanks so much for all your help. I tried Steve's suggestion using tail and found:from=[EMAIL PROTECTED] , ...stat=Deferred: 450 [EMAIL PROTECTED]: Sender address rejected: Domain not foundSo it looks like the sender email is no longer acceptable. This worked fine b4, so perhaps the ISP taking care of the callees email has changed its policy? Anyway, I could use a hand on how to fix this. How do I get Asterisk to use a valid email address rather than [EMAIL PROTECTED]?Many Thanks,Hugh On 3/17/06, Steve Jones [EMAIL PROTECTED] wrote: Do a tail -f /var/log/maillog which will give you areal-time view of your mail server activity, then while that's running, leave yourself a voicemail.From: Tony Mountifield [mailto:[EMAIL PROTECTED]]Sent: Fri 3/17/2006 10:13 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: problems with emailing voicemailIn article [EMAIL PROTECTED] ,hugolivude [EMAIL PROTECTED] wrote: I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0 .Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail.There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get Asterisk going, I'm hoping someone can steer me in the right direction! Any suggestions where/how to troubleshoot?The first place to look would be in /var/log/maillog on the box. Look particularly around the time when a voicemail should have been sent.CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation
Re: [Asterisk-Users] simple perl-agi - where's the error?
no. the result is sligthly different(no quotes), but the variable still can not be written: GET VARIABLE DIALSTRING AGI Tx 200 result=1 (Zap/G1/0892343242343) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: ((null)) Mar 20 16:12:10 WARNING[4478]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 i don't think the problem lies in the dial-command, but in setting the variable $dialstring when i use $dialstring = Zap/G1/0892343242343 the dial-command works... thanks! On Mon, 20 Mar 2006 15:06:47 +0100 Benoît Mérouze [EMAIL PROTECTED] wrote: You should try '$res = $AGI-exec(DIAL, $dialstring);' Christian B wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI-get_variable(DIALSTRING); $res = $AGI-exec(DIAL $dialstring); the asterisk output says: AGI Rx GET VARIABLE DIALSTRING AGI Tx 200 result=1 (089324154332) AGI Rx EXEC DIAL -- AGI Script Executing Application: (DIAL) Options: () Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires an argument (technology/number) AGI Tx 200 result=-1 -- AGI Script agirouter/dialscript.pl completed, returning 0 so the get_variable-command seems to work, also the exec(with $dialstring = 089324154332 the call goes out), but not setting the variable. should be so simple :-( astcc-agi seems to use the same syntax, so i have no clue what is wrong in my place... any ideas? thx! kind regards christian ___ --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 -- Benoit Merouze Network Software Developer [EMAIL PROTECTED] ___ --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 ___ --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] pickup problem
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 20, 2006 4:06 PM there is also a more generic call pickup using 'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach uses *8 or *8# to pickup any ringing phone within the callgroup number (eg, 2 in this example). Does this call pickup work with IAX2? If yes, how, if there is no callgroup/pickupgroup setting in iax.conf? More in general: does call pickup work between different protocols? Thanks Mimmus ___ --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] Re: problems with emailing voicemail
Maybe I am misunderstanding what you did here, but I just want to make sure... First, in the network' file, the goal was to change the hostname from localhost.localdomain reference to a real hostname that would be accepted, so that the file would look more like: NETWORKING=yes HOSTNAME=asterisk.mydomain.com And the other poster was recommeding that the hosts file, should be setup to point the loopback address to that name as well, such as adding (or modifying the existing 127.0.0.1 line to look like: 127.0.0.1 asterisk.mydomain.com Another thing to do, would be to go to your /etc/mail/access file [vi /etc/mail/access] ahd make sure, or add your localhost as a trusted machine for sendmail.. From this link http://www.linuxhomenetworking.com/linux-hn/sendmail.htm I am reading/recommending that you put lines such as these in your access file: localhost.localdomain RELAY localhost RELAY 127.0.0.1 RELAY If your access file isn't in the /etc/mail directory, you may have to do a locate access to find it.. locate will only work if you have sometime in the past run an updatedb to build the hard drive's index for the locate to work on it... Hope this helps.. -Steve From: hugolivude [mailto:[EMAIL PROTECTED] Sent: Mon 3/20/2006 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: problems with emailing voicemail Guys, Thanks again for all your help. I've updated /etc/sysconfig/network and /etc/hosts as per your suggestions: /etc/sysconfig/network: NETWORKING=yes HOSTNAME=localhost.localdomain 127.0.0.1 my external, static IP address asterisk localhost /etc/hosts: # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1localhost.localdomainlocalhost 127.0.0.1 my external, static IP address asterisk localhost But still no luck. I've also noticed a problem when I boot the server. Sendmail along with a number of other processes generate an errror message. They're all pretty much the same error message and same line number so here's an example: etc/rc.d/rc.sysinit error line 3: 127.0.0.1 command not found FWIW, I've included the first part of my etc/rc.d/rc.sysinit below. I'm stumped. This was working before, what could have possibly happened? Any other ideas on how to fix it before I go and re-install RedHat?? Thanks, Hugh #!/bin/bash # # /etc/rc.d/rc.sysinit - run once at boot time # # Taken in part from Miquel van Smoorenburg's bcheckrc. # # Rerun ourselves through initlog if [ -z $IN_INITLOG -a -x /sbin/initlog ]; then exec /sbin/initlog $INITLOG_ARGS -r /etc/rc.d/rc.sysinit fi # If we're using devfs, start devfsd now - we need the old device names [ -e /dev/.devfsd -a -x /sbin/devfsd ] /sbin/devfsd /dev HOSTNAME=`/bin/hostname` if [ -f /etc/sysconfig/network ]; then . /etc/sysconfig/network else NETWORKING=no fi if [ -z $HOSTNAME -o $HOSTNAME = (none) ]; then HOSTNAME=localhost fi . /etc/init.d/functions # Start the graphical boot, if necessary if [ $BOOTUP = graphical ]; then if [ -x /usr/bin/rhgb ]; then /usr/bin/rhgb else export BOOTUP=color fi fi last=0 for i in `LC_ALL=C grep '^[0-9]*.*respawn:/sbin/mingetty' /etc/inittab | sed 's/^.* tty\([0-9][0-9]*\).*/\1/g'`; do /dev/tty$i last=$i done if [ $last -gt 0 ]; then /dev/tty$((last+1)) /dev/tty$((last+2)) fi if [ `/sbin/consoletype` = vt -a -x /sbin/setsysfont ]; then echo -n Setting default font ($SYSFONT): /sbin/setsysfont if [ $? -eq 0 ]; then success else failure fi echo ; echo fi On 3/17/06, Colin Anderson [EMAIL PROTECTED] wrote: if you are using Sendmail, then you have to add a trusted user to /etc/sendmail.cf in the format: Tuser So if the user you run Asterisk under is called asterisk then you add Tasterisk to sendmail.cf. Restart Sendmail. Then in voicemail.conf you can set the email address to whatever you want, it's at the top of the config file, I believe. -Original Message- From: hugolivude [mailto:[EMAIL PROTECTED] Sent: Friday, March 17, 2006 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: problems with emailing voicemail Wow, Thanks so much for all your help. I tried Steve's suggestion using tail and found: from=[EMAIL PROTECTED] , ... stat=Deferred: 450 [EMAIL PROTECTED]: Sender address rejected: Domain not found So it looks like the sender email is no longer acceptable. This worked fine b4, so perhaps the ISP taking care of the callees email has
Re: [Asterisk-Users] pickup problem
Mimmus wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 20, 2006 4:06 PM there is also a more generic call pickup using 'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach uses *8 or *8# to pickup any ringing phone within the callgroup number (eg, 2 in this example). Does this call pickup work with IAX2? If yes, how, if there is no callgroup/pickupgroup setting in iax.conf? More in general: does call pickup work between different protocols? Never had a need to do pickup with iax, so don't have a clue. As I recall, the callgroup keyword only applies to sip and zap channels. ___ --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] pickup problem
On 20 Mar 2006, at 15:39, Rich Adamson wrote: Mimmus wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 20, 2006 4:06 PM there is also a more generic call pickup using 'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach uses *8 or *8# to pickup any ringing phone within the callgroup number (eg, 2 in this example). Does this call pickup work with IAX2? If yes, how, if there is no callgroup/pickupgroup setting in iax.conf? More in general: does call pickup work between different protocols? Never had a need to do pickup with iax, so don't have a clue. As I recall, the callgroup keyword only applies to sip and zap channels. It doesn't work between protocols. Tim Panton [EMAIL PROTECTED] ___ --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
[Asterisk-Users] How often do YOU register?
Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations? ___ --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] Zaptel will not build
Chris Mason (Lists) wrote: FYI: I am trying to build zaptel-1.2.4 against the recently updated kernel version 2.6.9-34.EL on Centos 4.2. but I am getting errors and it will not build. This is apparently due to a typo in a kernel header spinlock.h although I have not successfully modified the kernel and built zaptel against it yet. https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 This bug report has a typo as well. It should read: #define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED snipped This solution worked for me. Scroll down until you find rebuilding zaptel. http://nerdvittles.com/index.php?p=123 Thanks, Mike Clark ___ --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] How to make caller groups ???
You'll want to learn all about Channels and groups. You can try here: http://www.voip-info.org/wiki/view/Channels+and+Groups
.
I've assumed that you have 4 FXO modules (to support 4 external phone lines) and 4 FXS modules (to support 4 local extensions).
Essentially you'll need
to define groups in your ZAPATA.CONF file. I've provided an
example below (it also includes call groups and pick up groups - I've
set it so that any group can pick up a call ringing for someone else)
You'll also need contexts in EXTENSIONS.CONF
to control how your users dial. The entries in ZAPATA.CONF for
the (FXS) extensions will invoke the appropriate context in
EXTENSIONS.CONF so that the appropriate trunks are used. Check
out the Dialling a Group section of http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels for information on the various ways you can have Asterisk select a trunk from a group. I've used r below -
a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group).
Yours,
Hugh
Extensions.conf:
[globals]
BOSS_TRUNK=ZAP/r1
GROUPA_TRUNK=ZAP/r2
GROUPB_TRUNK=ZAP/r3
GROUPC_TRUNK=ZAP/r4
[boss-context]
;North
American Long Distance
exten =
_1XX,1,Dial(BOSS_TRUNK/${EXTEN})
[groupA-context]
;North
American Long Distance
exten =
_1XX,1,Dial(GROUPA_TRUNK/${EXTEN})
[groupB-context]
;North
American Long Distance
exten =
_1XX,1,Dial(GROUPB_TRUNK/${EXTEN})
[groupC-context]
;North
American Long Distance
exten =
_1XX,1,Dial(GROUPC_TRUNK/${EXTEN})
Zapata.conf:
;FXS Line 1
– The Boss's local extension
language=en
context= boss-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=1
pickupgroup=1,2,3,4
channel=1
;
;FXS Line 2
– Group A's local extension
language=en
context= groupA-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=2
pickupgroup=1,2,3,4
channel=1
;
;FXS Line 3
– Group B's local extension
language=en
context= groupB-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=3
pickupgroup=1,2,3,4
channel=1
;
;FXS Line 4
– Group C's local extension
language=en
context= groupC-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=4
pickupgroup=1,2,3,4
channel=1
;
;FXO
(incoming) Line 1
language=en
context=Boss-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=1
channel
= 1
;
;FXO
(incoming) Line 2
language=en
context=general-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=2
channel
= 1
;
;FXO
(incoming) Line 3
language=en
context=general-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=2,3
channel
= 1
;
;FXO
(incoming) Line 4
language=en
context=general-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=2,3,4
channel
= 1
On 3/20/06, Faisal Inam [EMAIL PROTECTED] wrote:
Hello All !!!
I have 4 PSTNlines in the PBX server 1,2,3,4.
Firstline will be usedby only one extension (i.e. for the boss) for incom
ing and
outgoing. This line is dedicated for him only.(
The remaining lines will be shared bythe employees
1) Group Ahave access to lines 2 , 3 4.
2)Group Bhave access tolines 3 4
3)Group C have access to line 4
I want to know that how i will make that groups.
I will be grateful for ur
help.
Thanks a lot. Faisal
Relax. Yahoo! Mail
virus scanning helps detect nasty viruses!
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Re: [Asterisk-Users] How to make caller groups ???
Woops, noticed that the channels in my example are channel =
1. You'll need to change that so it jibes with your ZAPTEL.CONF
file...
H
On 3/20/06, hugolivude [EMAIL PROTECTED] wrote:
You'll want to learn all about Channels and groups. You can try here:
http://www.voip-info.org/wiki/view/Channels+and+Groups
.
I've assumed that you have 4 FXO modules (to support 4 external phone lines) and 4 FXS modules (to support 4 local extensions).
Essentially you'll need
to define groups in your ZAPATA.CONF file. I've provided an
example below (it also includes call groups and pick up groups - I've
set it so that any group can pick up a call ringing for someone else)
You'll also need contexts in EXTENSIONS.CONF
to control how your users dial. The entries in ZAPATA.CONF for
the (FXS) extensions will invoke the appropriate context in
EXTENSIONS.CONF so that the appropriate trunks are used. Check
out the Dialling a Group section of http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels
for information on the various ways you can have Asterisk select a trunk from a group. I've used r below -
a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group).
Yours,
Hugh
Extensions.conf:
[globals]
BOSS_TRUNK=ZAP/r1
GROUPA_TRUNK=ZAP/r2
GROUPB_TRUNK=ZAP/r3
GROUPC_TRUNK=ZAP/r4
[boss-context]
;North
American Long Distance
exten =
_1XX,1,Dial(BOSS_TRUNK/${EXTEN})
[groupA-context]
;North
American Long Distance
exten =
_1XX,1,Dial(GROUPA_TRUNK/${EXTEN})
[groupB-context]
;North
American Long Distance
exten =
_1XX,1,Dial(GROUPB_TRUNK/${EXTEN})
[groupC-context]
;North
American Long Distance
exten =
_1XX,1,Dial(GROUPC_TRUNK/${EXTEN})
Zapata.conf:
;FXS Line 1
– The Boss's local extension
language=en
context= boss-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=1
pickupgroup=1,2,3,4
channel=1
;
;FXS Line 2
– Group A's local extension
language=en
context= groupA-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=2
pickupgroup=1,2,3,4
channel=1
;
;FXS Line 3
– Group B's local extension
language=en
context= groupB-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=3
pickupgroup=1,2,3,4
channel=1
;
;FXS Line 4
– Group C's local extension
language=en
context= groupC-context
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callgroup=4
pickupgroup=1,2,3,4
channel=1
;
;FXO
(incoming) Line 1
language=en
context=Boss-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=1
channel
= 1
;
;FXO
(incoming) Line 2
language=en
context=general-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=2
channel
= 1
;
;FXO
(incoming) Line 3
language=en
context=general-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=2,3
channel
= 1
;
;FXO
(incoming) Line 4
language=en
context=general-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
transfer=yes
rxgain=5%
group=2,3,4
channel
= 1
On 3/20/06, Faisal Inam
[EMAIL PROTECTED] wrote:
Hello All !!!
I have 4 PSTNlines in the PBX server 1,2,3,4.
Firstline will be usedby only one extension (i.e. for the boss) for incom
ing and
outgoing. This line is dedicated for him only.(
The remaining lines will be shared bythe employees
1) Group Ahave access to lines 2 , 3 4.
2)Group Bhave access tolines 3 4
3)Group C have access to line 4
I want to know that how i will make that groups.
I will be grateful for ur
help.
Thanks a lot. Faisal
Relax. Yahoo! Mail
virus scanning helps detect nasty viruses!
___--Bandwidth and Colocation provided by
Easynews.com
--Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
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Re: [Asterisk-Users] How often do YOU register?
Hi, Being SER user I use 5 minutes (300 seconds). But you have to balance between load on your registrar server (like * in this case) and keeping your database up to date. Too short re-registration in huge system means literally tens of registration per second. To long registration means: a) users will seem available for a prolonged time, even during power off, their network failure, etc. b) if user will change IP address and you have long expiratoin time, he will be visible under two or more addresses. Most SIP proxies will perform parallel forking - they will contact all IPs. Of coure provided that you do not perform any form of registrar database 'purge' or cleaning, when registering UA from new IP address Matt wrote: Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations? -- Regards, Arek Bekiersz ___ --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
[Asterisk-Users] Prodding channel h323
hello, sometimes per day, below messages appears in my asterisk/messages log... any suggestion, what this mean? thx PJ Mar 20 07:57:53 WARNING[4672] channel.c: Prodding channel 'H323/ip$172.20.1.11:53473/331' failed Mar 20 07:57:53 NOTICE[14280] chan_h323.c: Avoiding H.323 destory deadlock on ip$172.20.1.11:53473/331 Mar 20 07:57:53 NOTICE[14280] chan_h323.c: Avoiding H.323 destory deadlock on ip$172.20.1.11:53473/331 ___ --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] How often do YOU register?
Ok, If a user drops off (power failure, etc). I detect them in asterisk as going offline within about 2 minutes. However, registration is only happening once an hour. I have qualify=yes set in asterisk. On 3/20/06, Arek Bekiersz [EMAIL PROTECTED] wrote: Hi, Being SER user I use 5 minutes (300 seconds). But you have to balance between load on your registrar server (like * in this case) and keeping your database up to date. Too short re-registration in huge system means literally tens of registration per second. To long registration means: a) users will seem available for a prolonged time, even during power off, their network failure, etc. b) if user will change IP address and you have long expiratoin time, he will be visible under two or more addresses. Most SIP proxies will perform parallel forking - they will contact all IPs. Of coure provided that you do not perform any form of registrar database 'purge' or cleaning, when registering UA from new IP address Matt wrote: Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations? -- Regards, Arek Bekiersz ___ --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 ___ --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
[Asterisk-Users] simple question on asterisk
Hi, I am planning to deploy an asterisk installation but I need to convince a few managers that its a good idea. Theres something I don't quite understand though, I plan deploy a box on the end of 4 channel BRI ISDN and provide it an ADSL internet connection. Should a phone behind the asterisk PBX wish to call a VOIP phone number number, say an 0844 one from www.voip-user.org, would it send this automatically over the PSTN ISDN network or would it know to send the call over the internet. Would I need a SIP provider on the internet to forward the calls? I assume I would need some sort of directory service to know where to route the call. Thanks in advance, Mark ___ --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] How often do YOU register?
My thought was.. I wondered if having it register more often (Every 5 minutes) might help some users who experience intermitent 'no dial tone' and have to 'reboot their device' On 3/20/06, Arek Bekiersz [EMAIL PROTECTED] wrote: Hi, Being SER user I use 5 minutes (300 seconds). But you have to balance between load on your registrar server (like * in this case) and keeping your database up to date. Too short re-registration in huge system means literally tens of registration per second. To long registration means: a) users will seem available for a prolonged time, even during power off, their network failure, etc. b) if user will change IP address and you have long expiratoin time, he will be visible under two or more addresses. Most SIP proxies will perform parallel forking - they will contact all IPs. Of coure provided that you do not perform any form of registrar database 'purge' or cleaning, when registering UA from new IP address Matt wrote: Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations? -- Regards, Arek Bekiersz ___ --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 ___ --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
[Asterisk-Users] pickup a call in queue
Hello, We are faced with a problem concerning queues. When we have several calls in different queues, is there some sort of way to open a channel between a (sip-)phone and a SPECIFIC call in a queue using the Asterisk manager api? We would like to do this even when we are not a member of that specific queue. Thanks in advance for any suggestions! cheers ___ --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] simple question on asterisk
Its all about how you configure your dialplan. Asterisk doesn't know what a PSTN or VOIP phone number is. If you want all 08444 numbers to go through a certain trunk, then you set your dialplan up accordingly. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hayward Sent: Monday, March 20, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] simple question on asterisk Hi, I am planning to deploy an asterisk installation but I need to convince a few managers that its a good idea. Theres something I don't quite understand though, I plan deploy a box on the end of 4 channel BRI ISDN and provide it an ADSL internet connection. Should a phone behind the asterisk PBX wish to call a VOIP phone number number, say an 0844 one from www.voip-user.org, would it send this automatically over the PSTN ISDN network or would it know to send the call over the internet. Would I need a SIP provider on the internet to forward the calls? I assume I would need some sort of directory service to know where to route the call. Thanks in advance, Mark ___ --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 ___ --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
[Asterisk-Users] assman, the ncurses asterisk manager interface
The project is now soundly set at assman.sf.net with a few more updates committed to SVN. I have not released an official release yet since the package is still considered beta quality, but it's quite easy to check out the SVN. -- Sig Langehttp://www.signuts.net/ ___ --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
[Asterisk-Users] hunt groups
What I would like to do is exten = 1000,1,Dial(sip/1000)(zap/g1,97837560) exten= 1000,2,Voicemail(u1000) Basically a follow me app that rings numerous interfaces and allows me to answer or it to time out and go to vmail. I didnt include the time out here as I am hoping someone can tell me where that needs to be. I really dont want to make the caller ring one interface and then the other. Ideally I would be able to press pound after answering so that it didnt continue to ring the other interface. Most of the apps that I saw do this are basically the same as forwarding the extension, any system can do that and I know asterisk is better than that. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --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] answer delay
FaberK wrote: Hi guys, maybe youìve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller-telco(telco bill to the caller as soon as * answer)-asterisk Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call. I don't think this is possible to implement from the Asterisk end of things. jd ___ --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] Grabbing the billsec and duration after a hangup.
The reason for it being 0 is because as long as you sit on the h
extension the call is not yet done, therefore asterisk has no clue
what those valuse are. If you use the h extension then you are messing
up the CDR.
On 3/20/06, Mark Ackroyd [EMAIL PROTECTED] wrote:
Hello,
I am wondering if someone has got any ideas that can help solve this
problem.
I have a dial plan that you call into, and depending on certain conditions
it calls out on a number grabbed from a database.
Something like this :
exten = s,n,Do something
exten = s,n,Do something else
exten = s,n,Dial(ZAP/g1/${OUTBOUND},${timeout})
I need to log the time the person was connected to $(OUTBOUND) , these are
duration and billsec in the CDR's
So at hangup I do something like this.
exten = h,1,DeadAGI(cdr-
outlogger.php|${CDR(start)}|${OUTBOUND}|${CDR(channel)}|${CDR(duration)}|$
{CDR(billsec)}|${CDR(disposition)}|${CDR(accountcode)})
Trouble is duration and billsec are *ALWAYS* 0 (zero), as if they have not
been loaded with the values, even though the channel is hung up.
Anyone got any ideas on how I can access ${CDR(duration)} and
${CDR(billsec)} in the hangup extension?
Thanks, hope I explained that well enough.
Mark
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Re: [Asterisk-Users] Re: problems with emailing voicemail
OK!! That's not what I did I've gone back and changed things according to what you indicated, thanks for making it so simple to folow... The Asterisk box is on an internal network so instead of asterisk.mydomain.com I tried using our external fixed IP address. The error messages have disappeared, but I'm still not getting email. Just for fun I changed the email I'm using in voicemail.conf to my gmail account and it worked!! When I use our work email though, it doesn't work here's the error I see using the 'tail' command: Mar 20 11:50:19 69 sendmail[2609]: k2KGoJvC002609: from=[EMAIL PROTECTED], size=88081, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=[127.0.0.1] Mar 20 11:50:20 69 sendmail[2606]: k2KGoJD1002606: to=Hugh Oliver [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:01, mailer=relay, pri=30325, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (k2KGoJvC002609 Message accepted for delivery) Mar 20 11:50:20 69 sendmail[2611]: k2KGoJvC002609: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=30451, relay=mail.gesturetek.com. [64.41.126.140], dsn=5.6.0, stat=Data format error Mar 20 11:50:20 69 sendmail[2611]: k2KGoJvC002609: k2KGoKvC002611: DSN: Data format error Mar 20 11:50:20 69 sendmail[2611]: k2KGoKvC002611: to=[EMAIL PROTECTED], delay=00:00:00, xdelay=00:00:00, mailer=local, pri=119105, dsn=2.0.0, stat=Sent I appreciate your help. I've pasted my hosts, network access files below. Thanks, Hugh hosts: # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 my IP address in the form ###.###.###.### network: NETWORKING=yes HOSTNAME=my IP address in the form ###.###.###.### access: # Check the /usr/share/doc/sendmail/README.cf file for a description # of the format of this file. (search for access_db in that file) # The /usr/share/doc/sendmail/README.cf is part of the sendmail-doc # package. # # by default we allow relaying from localhost... localhost.localdomain RELAY localhost RELAY 127.0.0.1 RELAY On 3/20/06, Steve Jones [EMAIL PROTECTED] wrote: Maybe I am misunderstanding what you did here, but I just want to make sure...First, in the network' file, the goal was to change the hostname from localhost.localdomain reference to a real hostname that would be accepted, so that the file would look more like:NETWORKING=yesHOSTNAME=asterisk.mydomain.com And the other poster was recommeding that the hosts file, should be setup to point the loopback address to that name as well, such as adding (or modifying the existing 127.0.0.1 line to look like: 127.0.0.1 asterisk.mydomain.comAnother thing to do, would be to go to your /etc/mail/access file[vi /etc/mail/access] ahd make sure, or add your localhost as a trusted machine for sendmail..From this link http://www.linuxhomenetworking.com/linux-hn/sendmail.htm I am reading/recommending that you put lines such as these in your access file: localhost.localdomain RELAYlocalhost RELAY127.0.0.1 RELAYIf your access file isn't in the /etc/mail directory, you may have to do a locate access to find it..locate will only work if you have sometime in the past run an updatedb to build the hard drive's index for the locate to work on it...Hope this helps..-SteveFrom: hugolivude [mailto: [EMAIL PROTECTED] ]Sent: Mon 3/20/2006 10:14 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: problems with emailing voicemailGuys,Thanks again for all your help.I've updated /etc/sysconfig/network and /etc/hosts as per your suggestions: /etc/sysconfig/network:NETWORKING=yesHOSTNAME=localhost.localdomain127.0.0.1my external, static IP address asterisk localhost /etc/hosts:# Do not remove the following line, or various programs # that require network functionality will fail.127.0.0.1localhost.localdomainlocalhost 127.0.0.1my external, static IP address asterisk localhost But still no luck.I've also noticed a problem when I boot the server.Sendmail along with a number of other processes generate an errror message.They're all pretty much the same error message and same line number so here's an example:etc/rc.d/rc.sysinit error line 3: 127.0.0.1 command not found FWIW, I've included the first part of my etc/rc.d/rc.sysinit below. I'm stumped.This was working before, what could have possibly happened?Any other ideas on how to fix it before I go and re-install RedHat??Thanks,Hugh#!/bin/bash## /etc/rc.d/rc.sysinit - run once at boot time## Taken in part from Miquel van Smoorenburg's bcheckrc.## Rerun ourselves through initlog if [ -z $IN_INITLOG -a -x /sbin/initlog ]; thenexec /sbin/initlog $INITLOG_ARGS -r /etc/rc.d/rc.sysinitfi# If we're using devfs, start devfsd now - we need the old device names[ -e /dev/.devfsd -a -x /sbin/devfsd ] /sbin/devfsd /dev HOSTNAME=`/bin/hostname`if [ -f /etc/sysconfig/network ]; then.
[Asterisk-Users] meetme recording very loud
Hi. I tried to record a meetme conference using the r option -- using asterisk 1.2.4 and the volume is so loud it clips. Any way to fix this -- using the monitor the volume is generally fine. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] Re: problems with emailing voicemail
On Mon, 2006-03-20 at 11:59 -0500, hugolivude wrote: OK!! That's not what I did I've gone back and changed things according to what you indicated, thanks for making it so simple to folow... The Asterisk box is on an internal network so instead of asterisk.mydomain.com I tried using our external fixed IP address. The error messages have disappeared, but I'm still not getting email. Just for fun I changed the email I'm using in voicemail.conf to my gmail account and it worked!! When I use our work email though, it doesn't work here's the error I see using the 'tail' command: So what has changed is that somebody has reconfigured your work email server to reject mail from domains it can't resolve. It would also appear that your works email server is giving the DSN in some unknown format from the log file. Easiest way to fix this would be to speak to the admin of your works email server and get him to add the IP of your * server into the relay_allow (or equivalent) list. Rgds Pete ___ --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] answer delay
On Monday 20 March 2006 11:46, John Daragon wrote: Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call. Incorrect. I do this all the time with a PRI. You can't do this with POTS. Simply don't Answer() until you're ready to bill. You can send audio but you cannot hear them until you answer the call. exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this) exten = 5551234,n,Answer exten = 5551234,n,Playback(now-the-meter-is-running) exten = 5551234,n,Record(and-we-can-hear-you.gsm) -A. ___ --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] answer delay
Thanks a lot!!! Is exactly what I need to do. Send a message, before answer. Thanks to all! F.2006/3/20, Andrew Kohlsmith [EMAIL PROTECTED]: On Monday 20 March 2006 11:46, John Daragon wrote: Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call.Incorrect.I do this all the time with a PRI.You can't do this with POTS. Simply don't Answer() until you're ready to bill.You can send audio but youcannot hear them until you answer the call.exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this)exten = 5551234,n,Answer exten = 5551234,n,Playback(now-the-meter-is-running)exten = 5551234,n,Record(and-we-can-hear-you.gsm)-A.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --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] hunt groups
What I would like to do is… exten = 1000,1,Dial(sip/1000)(zap/g1,97837560) exten= 1000,2,Voicemail(u1000) Basically a follow me app that rings numerous interfaces and allows me to answer or it to time out and go to vmail. I didn’t include the time out here as I am hoping someone can tell me where that needs to be. I really don’t want to make the caller ring one interface and then the other. Ideally I would be able to press pound after answering so that it didn’t continue to ring the other interface. Most of the apps that I saw do this are basically the same as forwarding the extension, any system can do that and I know asterisk is better than that. Either put the Dial commands in sequence with a short timeout, or put multiple arguments to the dial command separated by Option 1) exten = 1000,1,Dial(SIP/1000|15) exten = 1000,2,Dial(Zap/g1,97837560|15) rings each extension for 15 seconds option 2) exten = 1000,1,Dial(SIP/1000Zip/g1,97837560) rings both extensions at oncefirst one to answer is the winner. ___ --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] NEWS: SIP Firmware Available for Cisco 7970
Anyone got this working yet?Nope :( Any update to this status? JR ___ --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
[Asterisk-Users] Aterisk with Realtime
Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled.Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us'Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior. Please use '_X.' instead at line 3Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 4 Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 5Mar 20 23:05:05 WARNING[2007] config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not availabl eMar 20 23:05:25 WARNING[2007] config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available ram ___ --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] Aterisk with Realtime
On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior. Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --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
[Asterisk-Users] Is it possible to turn off password for transfers on FOP
Hi, Is it possible to turn off the request for a security code when transferring in FOP (Flash Operator Panel)? If not can the security code be set to use the SIP or voicemail passwords? I know there is a forum for FOP but no one seems to be answering there... so I thought I would see if anyone here might have experience with FOP. Thanks ___ --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] pickup problem
PickUp2: http://linux.thorsten-knabe.de/asterisk/pickup.jsp works very well. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Monday, March 20, 2006 4:50 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] pickup problem On 20 Mar 2006, at 15:39, Rich Adamson wrote: Mimmus wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 20, 2006 4:06 PM there is also a more generic call pickup using 'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach uses *8 or *8# to pickup any ringing phone within the callgroup number (eg, 2 in this example). Does this call pickup work with IAX2? If yes, how, if there is no callgroup/pickupgroup setting in iax.conf? More in general: does call pickup work between different protocols? Never had a need to do pickup with iax, so don't have a clue. As I recall, the callgroup keyword only applies to sip and zap channels. It doesn't work between protocols. Tim Panton ___ --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
[Asterisk-Users] (no subject)
Hi everybody. Yesterday I fix typo in spinlock.h and compiled zaptel. But today I have problems with soft phones. I tried to recompile zaptel and it showed errors again. So I don't understand what now it needs. Brings words and photos together (easily) with PhotoMail - it's free and works with Yahoo! Mail.___ --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
[Asterisk-Users] sip show inuse not accurate
hi, the command sip show inuse is giving me wrong results , the outgoing column is not working, look at this, (i have an outgoing call on 22662848 and it appears free) asterisk*CLI sip show inuse UsernameincomingLimit outgoingLimit 226628490 N/A 0 N/A 226628480 N/A 0 N/A 226628471 N/A 0 N/A 226628460 N/A 0 N/A 226628451 N/A 0 N/A i have and two atas (linksys pap2) is this a known gub or something? --- miguel ___ --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] Aterisk with Realtime
That, and make sure you've got extconfig set to use mysql for it's sippusers and sippeers and not odbc. Aaron Patrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior. Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --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 ___ --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] answer delay
Andrew Kohlsmith wrote: On Monday 20 March 2006 11:46, John Daragon wrote: Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call. Incorrect. I do this all the time with a PRI. You can't do this with POTS. Simply don't Answer() until you're ready to bill. You can send audio but you cannot hear them until you answer the call. Hell, you learn something new every short period of time. I have to go try this out... jd ___ --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] answer delay
On Monday 20 March 2006 13:49, John Daragon wrote: Hell, you learn something new every short period of time. I have to go try this out... :-) It's called early audio in PRI parlance, some carriers do not offer it but almost all do. -A. ___ --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] Aterisk with Realtime
Hi thanks for the reply this what my extconfig sipusers = odbc,asterisk,2_Sipsippeers = odbc,asterisk,2_Sipextensions = odbc,asterisk,2_Extensionsvoicemail = odbc,asterisk,2_VMUsersvoicemail_messages = odbc,asterisk,2_VM waht is wrong with this ? ram On 3/20/06, Aaron Daniel [EMAIL PROTECTED] wrote: That, and make sure you've got extconfig set to use mysql for it'ssippusers and sippeers and not odbc. AaronPatrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior.Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --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___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
[Asterisk-Users] Problem with intermittent one-way audio
Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call again, it usually works. I have tried both with trunk=yes, and trunk=no but they are still having the problem. The debug log has a lot of the following, but not much else to go on. Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received iseqno 18 not within window 19-19 Any help much appreciated. -- -Barry Flanagan ___ --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] Problem with intermittent one-way audio
I am having this problem also. I have 2 systems running 1.2.5. I had the problem and one system was running 1.2.4 and the other was running a CVS HEAD from October so I upgraded them both to 1.2.5 with no success. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Barry Flanagan Sent: Monday, March 20, 2006 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem with intermittent one-way audio Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call again, it usually works. I have tried both with trunk=yes, and trunk=no but they are still having the problem. The debug log has a lot of the following, but not much else to go on. Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received iseqno 18 not within window 19-19 Any help much appreciated. -- -Barry Flanagan ___ --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 ___ --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
[Asterisk-Users] need to make my oh323 work with quintum no gatekeeper
Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY --IP ASTERISK (OH323) Both are gateways.. but I dont know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie ___ --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] Problem with intermittent one-way audio
Barry Flanagan wrote: Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. I've had this problem in the past, when not running the same version of Asterisk on both ends of the trunk. Doug ___ --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] Aterisk with Realtime
Since you say you're using mysql as the backend, you need to change anything that says odbc to mysql so that the server knows where to find the db at. Also, you need to make sure the DB info is in res_mysql.conf. Aaron ram wrote: Hi thanks for the reply this what my extconfig sipusers = odbc,asterisk,2_Sip sippeers = odbc,asterisk,2_Sip extensions = odbc,asterisk,2_Extensions voicemail = odbc,asterisk,2_VMUsers voicemail_messages = odbc,asterisk,2_VM waht is wrong with this ? ram On 3/20/06, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: That, and make sure you've got extconfig set to use mysql for it's sippusers and sippeers and not odbc. Aaron Patrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior. Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --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
[Asterisk-Users] Experiences with ATA model Octtel SP200SO
Hi, I took a piece of Octtel SP200SO (SIP, FXS, FXO, LAN, WAN, QoS ...) from local distributor for testing. First surprise came, when I opened it ... no documentation was included, the second, when I learned, there is no relevant doc on company web too! Nevertheless in the rest I configured it (with partial success) ... - both ports (FXS+FXO) registers to asterisk (type=friend) - call from any SIP phone to the analog phone connected to FXS and vice versa is possible - call from any SIP phone (or FXS) to PSTN (via FXO) is possible, but callee sound volume is insufficient (although ATA's speaking volume is on maximum) - incoming call to FXO port is NOT forwarded to asterisk (no one packet arrives) although it is registered as friend, but after circa 4-th ring FXS phone starts to ring. If I pick it up, call establishes randomly (mostly not) Is there somebody with positive experience using this ATA ? noro ___ --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
[Asterisk-Users] Primary D-Channel on span 1 down
Hello, I got a Problem with my HFC Card, I start my asterisk -c The console comes up: ---snip-- Asterisk Ready. *CLI ---snip-- Setting up debuglevel for span 1 ---snip-- *CLI pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 ---snip-- Console output: ---snip-- *CLI 1 Sending TEI Request ri=24393 1 [ fc ff 03 0f 5f 49 01 ff ] 1 Unnumbered frame: 1 SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 1M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data 1 Sending TEI Request ri=46665 1 [ fc ff 03 0f b6 49 01 ff ] 1 Unnumbered frame: 1 SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 1M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data 1 Sending TEI Request ri=49657 1 [ fc ff 03 0f c1 f9 01 ff ] 1 Unnumbered frame: 1 SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 1M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data == Primary D-Channel on span 1 down ---snip-- The D-Channel on span 1 goes down, and I don't know why. -/etc/zaptel.conf- loadzone=de defaultzone=de span=1,0,0,ccs,ami bchan=1-2 dchan=3 - ---output of ztcfg -v --- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) 3 channels configured. - Does anybody know an solution for this problem? MfG, Christian Reelfs -- nnGa Burgstraße 22 - 26723 Emden - Germany Tel: +49 4921/99 34 11 | mobile: +49 160/94 52 57 15 Tel2: +49 4461/ 83 12 0 E-Mail [EMAIL PROTECTED] - URL: http://chris.mynnga.de ICQ 85564186 - skype: chrisnnga [Disclaimer] Diese Information ist ausschliesslich fuer die adressierte Person oder Organisation bestimmt und koennte vertrauliches und/oder privilegiertes Material enthalten. Personen oder Organisationen, fuer die diese Information nicht bestimmt ist, ist es nicht gestattet, diese zu lesen, erneut zu uebertragen, zu verbreiten, anderweitig zu verwenden oder sich durch sie veranlasst zu sehen, Massnahmen irgendeiner Art zu ergreifen. Sollten Sie diese Nachricht irrtuemlich erhalten haben, bitten wir Sie, sich mit dem Absender in Verbindung zu setzen und das Material von Ihrem Computer zu loeschen. Wir weisen darauf hin, dass derartige Nachrichten mit und ohne Zutun von Dritten verloren gehen, veraendert oder verfaelscht werden koennen. Herkoemmliche E-Mails sind nicht gegen den Zugriff von Dritten geschuetzt und deshalb ist auch die Vertraulichkeit unter Umstaenden nicht gewahrt. Wir haften deshalb nicht fuer die Unversehrtheit von E-Mails nachdem sie unseren Herrschaftsbereich verlassen haben und koennen Ihnen hieraus entstehende Schaeden nicht ersetzen. Sollte trotz der von uns verwendeten Virus-Schutz-Programmen durch die Zusendung von E-Mails ein Virus in Ihre Systeme gelangen, haften wir nicht fuer eventuell hieraus entstehende Schaeden. Dieser Haftungsausschluss gilt nur soweit gesetzlich zulaessig. ___ --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] Grabbing the billsec and duration after a hangup.
The reason for it being 0 is because as long as you sit on the h extension the call is not yet done, therefore asterisk has no clue what those valuse are. If you use the h extension then you are messing up the CDR. So how can I tell it the call is complete and give the CDR values? Is it just not possible? Mark ___ --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] Feedback from VON expo! Info on * HAandPolycomphone!!
The only thing is I want to be sure I understand the statement above because the only time I can see Asterisk needing to do an SRV lookup is if it is handing a call to a carrier for termination. Gabe, that is what I was talking about. Asterisk really needs the ability to make use of the termination providers' SRV records for failover. When one doesn't respond... use the next record. This work exceptionally well when connecting to directly to a provider with an ATA that supports SRV. It is a shame that * doesn't do this. I have been tempted to grab the SRV lookup code from the Jabber project and try to merge it into Asterisk. http://www.jabberstudio.org/cgi-bin/viewcvs.cgi/cvs/jabberd/jads2s/srv.c?view=markup I kinda understand the Jabber code, but I really have no idea where to begin in Asterisk. regards, David ___ --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] Problem with intermittent one-way audio
Doug Lytle wrote: Barry Flanagan wrote: Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. I've had this problem in the past, when not running the same version of Asterisk on both ends of the trunk. Thanks, I'll upgrade the remote end and see if that helps. -- -Barry Flanagan ___ --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
[Asterisk-Users] integration with Toshiba PBX system
Hi, I am currently integrating our company's Toshiba PBX with the Asterisk version 1.2.1. I bought Quad T1 card, and making the port 1 to connect to PSTN PRI (use pri_cpe in zaptel.conf) and making the port 3 to connect to Toshiba PBX (using pri_net in zaptel.conf). The first stage goal is to just adding the Asterisk relay between PSTN and Toshiba system. The issue I am facing is that I can make a outgoging call from Toshiba system phones to outside; but incoming calls always fail. I can observer the call come from span 1 and routes to span 3, but the call immediatedly hangup. Did anyone have experience on this issue. I try to make use the setting of misdn.conf to try to print out the signalling info, but it seems that there is no logging output. Is misdn.conf useful in 1.2.1 version. best regards Charles ___ --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] Feedback from VON expo! Info on * HAandPolycomphone!!
Is it so difficult to add a line in the dialplan directly under the one that fails to failover to? Aaron David Thomas wrote: The only thing is I want to be sure I understand the statement above because the only time I can see Asterisk needing to do an SRV lookup is if it is handing a call to a carrier for termination. Gabe, that is what I was talking about. Asterisk really needs the ability to make use of the termination providers' SRV records for failover. When one doesn't respond... use the next record. This work exceptionally well when connecting to directly to a provider with an ATA that supports SRV. It is a shame that * doesn't do this. I have been tempted to grab the SRV lookup code from the Jabber project and try to merge it into Asterisk. http://www.jabberstudio.org/cgi-bin/viewcvs.cgi/cvs/jabberd/jads2s/srv.c?view=markup I kinda understand the Jabber code, but I really have no idea where to begin in Asterisk. regards, David ___ --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 ___ --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
[Asterisk-Users] Asterisk Disconnecting after 30sec when someone leaving VM
Hello, I have started having a strange problem. Asterisk is connected via 4 analog lines to PSTN and we have SIP phones internally. All was working fine but recently each time a user calls from PSTN and when he is leaving a voicemail for someone, the caller gets disconnected after 30 secs. We have AMP installed. This is reproducible and is happening always. It seems that Asterisk is disconnecting because of a silence of 10 secs. But that is not case because the caller is still leaving a message. When calling from a SIP to SIP phone, this issue does not occur. Here is a log: - -- Executing VoiceMail(Zap/3-1, u533) in new stack -- Playing '/var/spool/asterisk/voicemail/default/533/temp' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/533/INBOX/msg00 -- x=1, open writing: /var/spool/asterisk/voicemail/default/533/INBOX/msg00 -- x=2, open writing: /var/spool/asterisk/voicemail/default/533/INBOX/msg00 -- Recording automatically stopped after a silence of 10 seconds -- Playing 'auth-thankyou' (language 'en') -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'Zap/3-1' in m == Spawn extension (macro-exten-vm, s, 8) exited non-zero on 'Zap/3-1' in macr == Spawn extension (aa_1, 240, 1) exited non-zero on 'Zap/3-1' -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (aa_1, h, 1) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' -- Hungup 'Zap/1-1' --- Any pointers/suggestions will be appreciated. Dave __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Feedback from VON expo! Info on *HAandPolycomphone!!
I am sure that must lead to potential trouble having multiple dial() one after the other as a form of HA. Besides, that wouldn't work for my LCDial(). - Gabe - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 20, 2006 1:47 PM Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on *HAandPolycomphone!! Is it so difficult to add a line in the dialplan directly under the one that fails to failover to? Aaron David Thomas wrote: The only thing is I want to be sure I understand the statement above because the only time I can see Asterisk needing to do an SRV lookup is if it is handing a call to a carrier for termination. Gabe, that is what I was talking about. Asterisk really needs the ability to make use of the termination providers' SRV records for failover. When one doesn't respond... use the next record. This work exceptionally well when connecting to directly to a provider with an ATA that supports SRV. It is a shame that * doesn't do this. I have been tempted to grab the SRV lookup code from the Jabber project and try to merge it into Asterisk. http://www.jabberstudio.org/cgi-bin/viewcvs.cgi/cvs/jabberd/jads2s/srv.c?view=markup I kinda understand the Jabber code, but I really have no idea where to begin in Asterisk. regards, David ___ --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 ___ --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 ___ --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
