[asterisk-users] Bug or feature: comments in chan_dahdi.conf.sample
Hi, 1. From chan_dahdi.conf.sample (asterisk 1.6.1.18) : ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN (common in Europe) ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; ;switchtype=euroisdn At the same time, dahdi_genconf generates files in which there is: ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=1,11 context=remote switchtype = euroisdn signalling = bri_cpe channel = 1-2 context = default group = 63 Does switchtype parameter also apply to BRI ? If positive, should we change this ; Switchtype: Only used for PRI comment ? 2. In chan_dahdi.conf.sample, you can also read ; national: National ISDN 2 (default). Using dahdi_genconf and without changing any default, I can see that switchtype is defaulted to euroisdn. Should this Switchtype default comment be changed ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug or feature: cdr_odbc.conf.sample
Hello, From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample : ; ; cdr_odbc.conf ; ;[global] ;dsn=MySQL-test ;username=username ;password=password ;loguniqueid=yes ;dispositionstring=yes ;table=cdr ;cdr is default table name ;usegmtime=no ; set to yes to log in GMT Though, reading from https://issues.asterisk.org/view.php?id=15021, it seems that lines username= and password= in cdr_odbc.conf are not used anymore (the fields in res_odbc.conf are used instead). My question are : 1. Are those lines still used in 1.6.1.X ? 2. If those lines are not used anymore, would you think more appropriate to remove them from cdr_odbc.conf.sample ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug or feature: cdr_odbc.conf.sample
On Sunday 18 April 2010 04:10:11 Olivier wrote: From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample : ; ; cdr_odbc.conf ; ;[global] ;dsn=MySQL-test ;username=username ;password=password ;loguniqueid=yes ;dispositionstring=yes ;table=cdr ;cdr is default table name ;usegmtime=no ; set to yes to log in GMT Though, reading from https://issues.asterisk.org/view.php?id=15021, it seems that lines username= and password= in cdr_odbc.conf are not used anymore (the fields in res_odbc.conf are used instead). My question are : 1. Are those lines still used in 1.6.1.X ? No. 2. If those lines are not used anymore, would you think more appropriate to remove them from cdr_odbc.conf.sample ? Removed, as of revision 257770. Thank you for the reminder. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug or feature: SIP chanvars not overriden
Hello,
Using 1.6.2-rc5, my settings include:
[local-phone](!)
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
subscribecontext=subs
disallow=all
allow=alaw
t38pt_udptl=no
setvar=accountcode=foo
[168](local-phone)
defaultuser=168
secret=pass168
callerid=John Doe168
mailbox=168
setvar=longcid=01555
setvar=accountcode=bar
CLI sip show peer 168
...
Variables:
accountcode = bar
longcid = 01555
accountcode = foo
When running, ${SIPPEER(168,chanvar[accountcode])}) is valued to foo
(instead of bar).
Would you rate it as a feature ?
Regards
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Re: [asterisk-users] Bug or feature: SIP chanvars not overriden
On Wednesday 11 November 2009 14:23:31 Olivier wrote:
Hello,
Using 1.6.2-rc5, my settings include:
[local-phone](!)
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
subscribecontext=subs
disallow=all
allow=alaw
t38pt_udptl=no
setvar=accountcode=foo
[168](local-phone)
defaultuser=168
secret=pass168
callerid=John Doe168
mailbox=168
setvar=longcid=01555
setvar=accountcode=bar
CLI sip show peer 168
...
Variables:
accountcode = bar
longcid = 01555
accountcode = foo
When running, ${SIPPEER(168,chanvar[accountcode])}) is valued to foo
(instead of bar).
Would you rate it as a feature ?
Neither. It's a misunderstanding on your part of how this all works. The
equivalent of your entry above is the context:
[168]
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
subscribecontext=subs
disallow=all
allow=alaw
t38pt_udptl=no
setvar=accountcode=foo
defaultuser=168
secret=pass168
callerid=John Doe168
mailbox=168
setvar=longcid=01555
setvar=accountcode=bar
The FIRST value is the value which takes precedence, not the last.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] Bug or feature : how to customize SIP REFER from dialplan
Hi, I've been editing my dialplan to launch custom instructions anytime a SIP REFER-based transfer occurs. The only hook I could find is catching an hangup event which is tied to a Zombie channel (ie a channel named like SIP/1234-vhvebjvnvZOMBIE). Is this a feature or a bug ? In other words, do you think : - it shouldn't be possible at all to hook custom instructions for SIP REFER-based transfer occurs (then I obviously found a bug), - catching ZOMBIE channel hangup is the way to hook custom instructions for a SIP REFER-based transfer. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?
I filed https://issues.asterisk.org/view.php?id=15202 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?
Hi, Digging on this case : 2009/5/26 Olivier oza-4...@myamail.com Hi, In my sip.conf, I've got : [general](+) ; register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 register=trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 When I'm using the TCP line instead of the other, I've got : [May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169 sip_parse_host: '/' is not a valid port number on line 25 of sip.conf. using default. [May 26 17:58:42] WARNING[2859]: chan_sip.c:6560 sip_register: Format for registration is [transport://]user[:secret[:authuse...@domain[:port][/extension][~expiry] at line 25 Is this register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 statement correct ? Regards I read in chan_sip.c that block inside sip_register : /* split [/contact][~expiry] */ expire = strchr(buf, '~'); if (expire) *expire++ = '\0'; callback = strrchr(buf, '/');// My comment: contact is search at the end of input register line if (callback) *callback++ = '\0'; if (ast_strlen_zero(callback)) callback = s; sip_parse_host(buf, lineno, username, portnum, transport); Given an input line such as register=tcp:// trunk4ipbx:passw...@192.168.100.129 trunk4ipbx%3apassw...@192.168.100.129, register line is truncated as the last occurence of '/' is the tcp:// string. When commenting out this callback = strrchr(buf, '/'); , input line register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 seems to be processed appropriately. My question is is this legal to input register lines without any /contact field ? If positive, then there is a bug is 1.6.1. If negative, would you agree to have a more appropriate logging than sip_parse_host: '/' is not a valid port number ... ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
At 7:33 AM -0700 on 5/26/04, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? [catching up on 800 -user posts - sorry for delay] Nobody on the list suggested this method that I saw: Use the Background application, but play silence. You'll notice in the asterisk-sounds directory (the additional package) there is a directory called silence which contains 10 files ranging from 1 to 10 seconds of silence. Works the same as Wait from the user's perspective (they hear nothing) but lets the user type keys. That's why I made those files; it's only a slightly ugly hack, and it works quite well. :-) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
John Todd wrote: At 7:33 AM -0700 on 5/26/04, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? [catching up on 800 -user posts - sorry for delay] Nobody on the list suggested this method that I saw: Use the Background application, but play silence. You'll notice in the asterisk-sounds directory (the additional package) there is a directory called silence which contains 10 files ranging from 1 to 10 seconds of silence. Works the same as Wait from the user's perspective (they hear nothing) but lets the user type keys. That's why I made those files; it's only a slightly ugly hack, and it works quite well. :-) Only slighly ? :-) -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bug or feature?
I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
On Wed, 2004-05-26 at 09:33, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? Wait() shouldn't take dtmf. It does seem odd till you realize that that is why there is timeouts and a timeout extension. Basically, if you are awaiting information from a user, just end your current priority and allow the timeout in your context to work. As for gotoif(), if you are processing dtmf and you start into a extension, asterisk has determined a match existed and is following your instructions. Only when you hit a point where you aren't telling asterisk what to do should it start listening for DTMF again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
On Wed, 26 May 2004, Maveric waxed: I've noticed that when i pass a wait in an exten = that it doesn't allow Are you talking about the Wait() application ? 'show application wait' for dtmf tone input. Also on another note i've noticed that when using Background() is what you want if you want to *wait* for DTMF. gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? GotoIf should execute a lot faster than your fingers can push buttons to send DTMF. Can you post the relevant section(s) of your extensions.conf ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
At 08:08 AM 5/26/2004, you wrote:
On Wed, 26 May 2004, Maveric waxed:
I've noticed that when i pass a wait in an exten = that it doesn't allow
Are you talking about the Wait() application ?
'show application wait'
for dtmf tone input. Also on another note i've noticed that when using
Background() is what you want if you want to *wait* for DTMF.
gotoif it will also cut the dtmf tones and drop the first part if the
gotoif is hit in the middle of input. Anybody else seen this or have this
problem?
GotoIf should execute a lot faster than your fingers can
push buttons to send DTMF. Can you post the relevant
section(s) of your extensions.conf ?
--Chris
--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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[easynews]
exten = s,1,SetVar,COUNTER=0;
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,BackGround(${SOUNDSDIR}/thank-you-for-calling-easynews)
exten = s,6,BackGround(${SOUNDSDIR}/new-direct-dial-number)
exten = s,7,BackGround(for-billing)
exten = s,8,BackGround(vm-press)
exten = s,9,BackGround(digits/1)
exten = s,10,BackGround(${SOUNDSDIR}/new-customer-signups)
exten = s,11,BackGround(vm-press)
exten = s,12,BackGround(digits/2)
exten = s,13,BackGround(for-tech-support)
exten = s,14,BackGround(vm-press)
exten = s,15,BackGround(digits/3)
exten = s,16,BackGround(${SOUNDSDIR}/if-you-know-your-partys-extension)
exten = s,17,BackGround(vm-press)
exten = s,18,BackGround(digits/5)
exten = s,19,SetVar,COUNTER=$[${COUNTER} + 1];
exten = s,20,GotoIf,$[${COUNTER} 4]?23:21
exten = s,21,Playback(vm-goodbye)
exten = s,22,Hangup
exten = s,23,Wait(0)
exten = 1,1,Goto(from-sip,1500,1)
exten = 2,1,Goto(from-sip,1505,1)
exten = 3,1,Goto(from-sip,1510,1)
exten = 5,1,Goto(ext-dial,s,1)
exten = 0,1,Goto(from-sip,1550,1)
exten = t,1,Goto(s,5)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,5)
It was much different before but this is how i worked around it. Also i
was calling Background before the wait but i think the wait should still
allow input.
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Re: [Asterisk-Users] bug or feature?
On Wed, 2004-05-26 at 14:45, Maveric wrote:
At 08:08 AM 5/26/2004, you wrote:
On Wed, 26 May 2004, Maveric waxed:
I've noticed that when i pass a wait in an exten = that it doesn't allow
Are you talking about the Wait() application ?
'show application wait'
for dtmf tone input. Also on another note i've noticed that when using
Background() is what you want if you want to *wait* for DTMF.
gotoif it will also cut the dtmf tones and drop the first part if the
gotoif is hit in the middle of input. Anybody else seen this or have this
problem?
GotoIf should execute a lot faster than your fingers can
push buttons to send DTMF. Can you post the relevant
section(s) of your extensions.conf ?
--Chris
--
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cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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[easynews]
exten = s,1,SetVar,COUNTER=0;
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,18,BackGround(digits/5)
exten = s,19,SetVar,COUNTER=$[${COUNTER} + 1];
exten = s,20,GotoIf,$[${COUNTER} 4]?23:21
exten = s,21,Playback(vm-goodbye)
exten = s,22,Hangup
exten = s,23,Wait(0)
Everything through s,19 is good. Delete s,20-23. Make a t extension so
you can wait a moment for input.
exten = t,1,GotoIf($[${COUNTER} 4]?s,5:h)
exten = h,1,Playback(vm-goodbye)
exten = h,2,Hangup()
exten = t,1,Goto(s,5)
You can't reach the t extension if you don't let asterisk fall out of
your commands into an idle state. That is why I mention how to fix it
the way I did above.
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,5)
It was much different before but this is how i worked around it. Also i
was calling Background before the wait but i think the wait should still
allow input.
Wait doesn't accept input, you need to quit telling asterisk to be
active and let it do it's work.
--
Steven Critchfield [EMAIL PROTECTED]
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