Re: [asterisk-users] gsm codec compile
oh, ok. I'm using asterisk-12. Native compiling On 4 March 2014 08:48, Doug dsc3...@yahoo.com wrote: Julian, The only thing I can thik of is that your source is different version. My GSM files are dated oct 19, 2008. There have been changes in the GSM Makefile over the years. Also are you compiling natively or cross compiling. I am compiling natively. Doug On Tuesday, March 4, 2014 12:54 AM, Julian Lyndon-Smith aster...@dotr.com wrote: this is all very odd. I have been compiling on raspbian wheezy for a few months now, and have never come across this error -rw-r--r-- 1 root root 6128 Aug 25 2013 codec_gsm.c -rw-r--r-- 1 root root126 Feb 8 08:50 codec_gsm.exports -rw-r--r-- 1 root root 181808 Feb 8 08:50 codec_gsm.o I don't have PROC defined in makeopts.in uname -a Linux hash42pi 3.10.25+ #622 PREEMPT Fri Jan 3 18:41:00 GMT 2014 armv6l GNU/Linux What could be causing my setup to compile ok, but not yours ? Julian On 4 March 2014 01:10, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: what about editing makeopts.in to have PROC=arm ? On Mon, Mar 3, 2014 at 9:48 PM, Doug dsc3...@yahoo.com wrote: OK set PROC to - OPTIMIZE+=-march=arm and now getting tons of errors in k6opt.s src/k6opt.s:350: Error: bad instruction `psllw %mm3,%mm0' src/k6opt.s:351: Error: bad instruction `movd %mm0,%eax' src/k6opt.s:352: Error: selected processor does not support ARM mode `movw %ax,(%esi)' src/k6opt.s:356: Error: bad instruction `emms' src/k6opt.s:358: Error: bad instruction `popl %esi' src/k6opt.s:359: Error: bad instruction `leave' src/k6opt.s:360: Error: bad instruction `ret' src/k6opt.s:367: Error: unrecognized symbol type src/k6opt.s:372: Error: unrecognized symbol type src/k6opt.s:382: Error: unrecognized symbol type src/k6opt.s:384: Error: bad instruction `pushl %ebp' src/k6opt.s:385: Error: bad instruction `movl %esp,%ebp' src/k6opt.s:386: Error: bad instruction `pushl %esi' Doug Crompton WA3DSP http://www.crompton.com On Monday, March 3, 2014 3:18 PM, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: Try to set PROC to arm. On Mon, Mar 3, 2014 at 7:13 PM, Doug dsc3...@yahoo.com wrote: I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it does not recognize the processor since it is one that is mentioned in the Makefile. Any help would be appreciated. make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' [AS] src/k6opt.s - src/k6opt.o Assembler messages: Error: unknown architecture `armv6l' Error: unrecognized option -march=armv6l make[2]: *** [src/k6opt.o] Error 1 Here are the lines in the Makefile - ifeq (, $(findstring $(OSARCH) , Darwin SunOS )) ifeq (, $(findstring $(PROC) , x86_64 amd64 ultrasparc sparc64 arm armv5b arm5b armeb hppa2.0 ppc powerpc ppc64 ia64 s390 bfin mipsel mips)) ifeq (, $(findstring $(shell uname -m) , ppc ppc64 alpha armv4l arm5b armv5b armv61 armv7l s390 )) OPTIMIZE+=-march=$(PROC) endif endif endif gcc is - Thread model: posix gcc version 4.6.3 (Debian 4.6.3-14+rpi1) # uname -m armv6l Doug -- _ -- 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 -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don't care if it works on your machine! We are not shipping your machine! The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- Julian Lyndon-Smith IT Director, Dot R Limited I don't care if it works on your machine! We are not shipping your machine! The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] gsm codec compile
this is all very odd. I have been compiling on raspbian wheezy for a few months now, and have never come across this error -rw-r--r-- 1 root root 6128 Aug 25 2013 codec_gsm.c -rw-r--r-- 1 root root126 Feb 8 08:50 codec_gsm.exports -rw-r--r-- 1 root root 181808 Feb 8 08:50 codec_gsm.o I don't have PROC defined in makeopts.in uname -a Linux hash42pi 3.10.25+ #622 PREEMPT Fri Jan 3 18:41:00 GMT 2014 armv6l GNU/Linux What could be causing my setup to compile ok, but not yours ? Julian On 4 March 2014 01:10, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: what about editing makeopts.in to have PROC=arm ? On Mon, Mar 3, 2014 at 9:48 PM, Doug dsc3...@yahoo.com wrote: OK set PROC to - OPTIMIZE+=-march=arm and now getting tons of errors in k6opt.s src/k6opt.s:350: Error: bad instruction `psllw %mm3,%mm0' src/k6opt.s:351: Error: bad instruction `movd %mm0,%eax' src/k6opt.s:352: Error: selected processor does not support ARM mode `movw %ax,(%esi)' src/k6opt.s:356: Error: bad instruction `emms' src/k6opt.s:358: Error: bad instruction `popl %esi' src/k6opt.s:359: Error: bad instruction `leave' src/k6opt.s:360: Error: bad instruction `ret' src/k6opt.s:367: Error: unrecognized symbol type src/k6opt.s:372: Error: unrecognized symbol type src/k6opt.s:382: Error: unrecognized symbol type src/k6opt.s:384: Error: bad instruction `pushl %ebp' src/k6opt.s:385: Error: bad instruction `movl %esp,%ebp' src/k6opt.s:386: Error: bad instruction `pushl %esi' Doug Crompton WA3DSP http://www.crompton.com On Monday, March 3, 2014 3:18 PM, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: Try to set PROC to arm. On Mon, Mar 3, 2014 at 7:13 PM, Doug dsc3...@yahoo.com wrote: I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it does not recognize the processor since it is one that is mentioned in the Makefile. Any help would be appreciated. make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' [AS] src/k6opt.s - src/k6opt.o Assembler messages: Error: unknown architecture `armv6l' Error: unrecognized option -march=armv6l make[2]: *** [src/k6opt.o] Error 1 Here are the lines in the Makefile - ifeq (, $(findstring $(OSARCH) , Darwin SunOS )) ifeq (, $(findstring $(PROC) , x86_64 amd64 ultrasparc sparc64 arm armv5b arm5b armeb hppa2.0 ppc powerpc ppc64 ia64 s390 bfin mipsel mips)) ifeq (, $(findstring $(shell uname -m) , ppc ppc64 alpha armv4l arm5b armv5b armv61 armv7l s390 )) OPTIMIZE+=-march=$(PROC) endif endif endif gcc is - Thread model: posix gcc version 4.6.3 (Debian 4.6.3-14+rpi1) # uname -m armv6l Doug -- _ -- 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 -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don't care if it works on your machine! We are not shipping your machine! The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Voice analytics
Does anyone know of a realtime voice analytic engine that works with asterisk 11+ ? We want to be able to listen on the conversation for key words in order to ensure compliance . The plan is to show these keywords onscreen, and remove them once the agent has covered the compliance issues. This would necessitate that the conversation is monitored and analysed in realtime as we can't do it post-call ;) Thanks Julian -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Asterisk 11 -CDR values changed in hangup handler not saved ?
check out the endbeforehexten option in cdr.conf this needs to set to yes Julian On 28 March 2013 23:56, Olivier oza_4...@yahoo.fr wrote: Hello, I'm using Hanhup Handlers in a testing asterisk 11 system. Within one such handler, I'm setting CDR values. To me, it seems those changed CDR values are not saved in CDR back-end. Can you confirm ? 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
Ah, right. Have a look at this documentation:
You may need to add some mapping
Julian
cdr_custom
This CDR backend allows for custom formatting of CDR records in a log file.
This module is most commonly used for customized CSV output. The
configuration file used for this module is /etc/asterisk/cdr_custom.conf. A
single section called [mappings] should exist in this file. The
[mappings] section
contains mappings between a filename and the custom template for a CDR. The
template is specified using Asterisk dialplan functions.
The following example shows a sample configuration for cdr_custom that
enables a single CDR log file, Master.csv. This file will be created as
/var/log/asterisk/cdr-custom/Master.csv. The template that has been defined
uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function
retrieves values from the CDR being logged. The CSV_QUOTE() function
ensures that the values are properly escaped for the CSV file format:
[mappings]
Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}
In the actual configuration file, the value in the Master.csv mapping
should be on a single line.
cdr_manager
On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
check out the endbeforehexten option in cdr.conf
this needs to set to yes
Julian
Unfortunately, this doesn't help.
Let's drop the hangup handler at the moment, and focus on the saving to
file part.
Then my issue is I can't update CDR value is hangup exten.
Here is a dialplan that illustrate this:
[from-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and
${CALLERID(num)})
same = n, Set(CDR(userfield)=foo)
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Set(CDR(userfield)=bar)
same = n, Hangup()
exten = h,1,Verbose(0,Entering context ${CONTEXT} from
${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
${EXTEN} and ${CALLERID(num)})
same = n,
ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)
My goal is to get either baz1 or baz2 value in
/var/log/asterisk/cdr-csv/Master.csv.
Typing channel originate Local/7005@from-foobar application Playback
tt-monkeys, I can see that the line with ExecIf is run but CDR still
contains foo value (the one set before Dial).
The strange thing is :
1. a CDR is written at the moment extension 7005 answers,
2. no other CDR is added when 7005 hangs up (so can't tell how long
extension 7005 listened to monkeys fellows).
(Setting endbeforehexten to either yes or no has no effect on this
behaviour.
My question are:
1. Is it simply possible to update CDR in hangup exten ?
2. How can I have a CDR for the application Playback part (see above) ?
3. Any tip or suggestion ?
Cheers
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your
machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
--
_
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Re: [asterisk-users] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
you can't set values in the h extension _unless_ you use the
endbeforehexten option in cdr.conf
you may need to reload the cdr module or restart asterisk for the
option to take effect.
It works. I know it does, as I use hangup handlers all the time. Much
better than the h extension ;)
Julian
On 29 March 2013 14:06, Olivier oza_4...@yahoo.fr wrote:
Thanks but I willingly choose a standard CDR field (I checked with both
accountcode and userfield) which appears in
/var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv away to
simplify things)
the fact found in Master.csv is foo, the value set before entering the
hangup extension (see previous dialplan)).
To me, this is either a feature (you can't set CDR values in hangup exten)
or a bug.
How would you qualify this ?
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
Ah, right. Have a look at this documentation:
You may need to add some mapping
Julian
cdr_custom
This CDR backend allows for custom formatting of CDR records in a log
file. This module is most commonly used for customized CSV output. The
configuration file used for this module is /etc/asterisk/cdr_custom.conf. A
single section called [mappings] should exist in this file. The [mappings]
section contains mappings between a filename and the custom template for a
CDR. The template is specified using Asterisk dialplan functions.
The following example shows a sample configuration for cdr_custom that
enables a single CDR log file, Master.csv. This file will be created as
/var/log/asterisk/cdr-custom/Master.csv. The template that has been defined
uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function
retrieves values from the CDR being logged. The CSV_QUOTE() function ensures
that the values are properly escaped for the CSV file format:
[mappings]
Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}
In the actual configuration file, the value in the Master.csv mapping
should be on a single line.
cdr_manager
On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
check out the endbeforehexten option in cdr.conf
this needs to set to yes
Julian
Unfortunately, this doesn't help.
Let's drop the hangup handler at the moment, and focus on the saving to
file part.
Then my issue is I can't update CDR value is hangup exten.
Here is a dialplan that illustrate this:
[from-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and
${CALLERID(num)})
same = n, Set(CDR(userfield)=foo)
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Set(CDR(userfield)=bar)
same = n, Hangup()
exten = h,1,Verbose(0,Entering context ${CONTEXT} from
${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
${EXTEN} and ${CALLERID(num)})
same = n,
ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)
My goal is to get either baz1 or baz2 value in
/var/log/asterisk/cdr-csv/Master.csv.
Typing channel originate Local/7005@from-foobar application Playback
tt-monkeys, I can see that the line with ExecIf is run but CDR still
contains foo value (the one set before Dial).
The strange thing is :
1. a CDR is written at the moment extension 7005 answers,
2. no other CDR is added when 7005 hangs up (so can't tell how long
extension 7005 listened to monkeys fellows).
(Setting endbeforehexten to either yes or no has no effect on this
behaviour.
My question are:
1. Is it simply possible to update CDR in hangup exten ?
2. How can I have a CDR for the application Playback part (see above) ?
3. Any tip or suggestion ?
Cheers
--
_
-- 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
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your
machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
--
_
-- Bandwidth and Colocation
Re: [asterisk-users] cisco 7940 and asterisk 11
wow. ok. um. Could I trouble you to send me the sip.conf, SIPDefault and Sip[mac] config files (with obvious things removed like peer name/password etc) - we upgraded from pre1-4 and I'm wondering if some of the settings are bad / simply wrong and would like to make a comparison. I know that it's a big ask, and I will understand if you decline to do so, but would really appreciate it if you could. Many thanks for the insight, either way. julian On 14 February 2013 15:27, Jeremy Kister asterisk...@jeremykister.comwrote: On 2/14/2013 1:20 AM, Julian Lyndon-Smith wrote: this is a real issue for us - anyone got _any_ clues or ideas ? Ever since we upgraded to asterisk 11 we have had audio problems with our cisco 7940 phones. I use all 7940 with my asterisk 1.8 upgraded to asterisk 11. I havent had any issues with call quality whatsoever. i'm running sip image 03-08-12 g711ulaw only. -- Jeremy Kister http://jeremy.kister.net./ -- Jeremy Kister http://jeremy.kister.net./ -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] cisco 7940 and asterisk 11
very polite *bump* this is a real issue for us - anyone got _any_ clues or ideas ? Thanks ;) On 12 February 2013 14:29, Julian Lyndon-Smith aster...@dotr.com wrote: Ever since we upgraded to asterisk 11 we have had audio problems with our cisco 7940 phones. The problems manifest themselves by the conversation turning robotic or into silence (to the extent our agents are saying hello? hello? and the customer is saying I hear you just fine We had to change pedantic=no in sip.conf to allow the phones to register We are assuming that it is the phone=asterisk combination because a) the call recordings of the conversation are perfect (no noise on the line, conversation is clear) but it is apparent that the agent cannot hear the customer sometimes (Hello?) b) we have replaced the cables and switches between the phones and the pbx c) we don't have the same problem with Aastra 9133i or Polycom 331 phones Are there any settings in sip.conf that may help this , or a particular firmware ? Are there any known audio problems with cisco 7940 and asterisk 11 ? Many thanks Julian -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] cisco 7940 and asterisk 11
Ever since we upgraded to asterisk 11 we have had audio problems with our cisco 7940 phones. The problems manifest themselves by the conversation turning robotic or into silence (to the extent our agents are saying hello? hello? and the customer is saying I hear you just fine We had to change pedantic=no in sip.conf to allow the phones to register We are assuming that it is the phone=asterisk combination because a) the call recordings of the conversation are perfect (no noise on the line, conversation is clear) but it is apparent that the agent cannot hear the customer sometimes (Hello?) b) we have replaced the cables and switches between the phones and the pbx c) we don't have the same problem with Aastra 9133i or Polycom 331 phones Are there any settings in sip.conf that may help this , or a particular firmware ? Are there any known audio problems with cisco 7940 and asterisk 11 ? Many thanks Julian -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Your thoughts and opinions on Asterisk 11 for production use
are you using cisco 79xx phones ? We had a similar problem. Upgrading the sip firmare to 8.12 fixed it for us. FWIW we're using 11 in a call centre, with 25k+ call attempts per day. Rock solid. Not a single crash since Oct 15 Julian On 10 January 2013 14:25, Christopher Harrington ch...@acsdi.com wrote: On Thu, Jan 10, 2013 at 8:18 AM, Danny Nicholas da...@debsinc.com wrote: I don’t presently have 11 in production, but in each case where I’ve put 11 in on top of 10.X the process has been relatively seamless, so I expect my 10.X boxes will go to 11.X sometime this year. Upgrading from 10.x to 11.x silently broke phone-to-phone call transfers for us. Haven't had time to investigate it more thoroughly, so we rolled back to 10. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Managing complex setups with Asterisk
A newbie question : Can each LXC client have their own ip address ? Thanks Julian On 8 November 2012 07:17, Olivier oza_4...@yahoo.fr wrote: 2012/11/7 Jeff LaCoursiere j...@sunfone.com Just to chime in, if you REALLY want multi-tenant, it is super easy and surprisingly efficient to use kernel level virtualization to run multiple instances of asterisk (and even FreePBX). We use LXC to do this. The host runs an instance that has the dahdi hardware, drivers, and upstream connections. The clients have SIP connections to the host for all inbound/outbound, so you have a central place to collect/process CDR records for billing. Getting your phones to connect to each instance is an exercise for the network admin ;) Much simpler than working out multiple contexts, extension overlaps, etc., IMO. Yes, it's a very interesting idea !! -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Random crash of the machine ? due to Asterisk 11
are you running dahdi ? We're using 11, System uptime: 3 weeks, 22 hours, 42 minutes, 19 seconds, 231452 calls processed We did, however, have a problem with dahdi freezing the machine Julian On 7 November 2012 22:32, asterisk asterisk aster...@ck-lee.com wrote: I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Random crash of the machine ? due to Asterisk 11
No, we removed dahdi (some hardware issues) and not had a problem since. No idea on the ldap side (you never mentioned ldap at all) On 8 November 2012 06:22, Samira Hosseini samiramhosse...@yahoo.com wrote: Hello, thanks for your reply. No , the daddi is not running on my asterisk server, Do you think it is necessary ? and the problem on LDAP is associate with dahdi? From: Julian Lyndon-Smith aster...@dotr.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, 8 November 2012, 9:40:50 Subject: Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11 are you running dahdi ? We're using 11, System uptime: 3 weeks, 22 hours, 42 minutes, 19 seconds, 231452 calls processed We did, however, have a problem with dahdi freezing the machine Julian On 7 November 2012 22:32, asterisk asterisk aster...@ck-lee.com wrote: I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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 -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Counting calls in progress from AMI
core show calls works for me on asterisk-11 I know it's a poll, but it's a very simple call ;) On 18 October 2012 16:28, Mitul Limbani mi...@enterux.in wrote: I guess you are looking for event handler, which can be polled programatically n not via manual command entry? Mitul On Oct 18, 2012 8:53 PM, Danny Nicholas da...@debsinc.com wrote: The AMI Command function issues CLI commands, but carry on. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 10:20 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI I need to do this from the AMI (not the CLI)...I don't *think* a comparable command exists from the AMI. As well, I don't want to poll the system for calls so I'm hoping to trap a call bridged,unbridged type event. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Thursday, October 18, 2012 10:59 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI The simplest way to accurately do this would be to issue command “core show channels verbose” From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 9:58 AM To: Asterisk Users List Subject: [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... Ideas? -- _ -- 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 -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] groups and categories
I know that I should know this. But I'm having serious brain farts at the
moment.
I want to have a call be counted in a number of ways
outbound
inbound
potential
so, for example, a call comes into my dialplan, I want to add it to
TotalCalls (all calls inbound + outbound)
InboundCalls (all inbound calls)
Potenial (all inbound calls that may potentially enter a conference)
Potenial (all inbound calls that may potentially enter a conference by
conference name)
the potential category needs to be decremented when the call actually
enters a conference
so, I have
exten = _[0-9A-Za-z].,n,Set(GROUP(conference)=TotalCalls)
exten = _[0-9A-Za-z].,n,Set(GROUP(conference)=Inbound)
exten = _[0-9A-Za-z].,n,Set(GROUP(conference)= Potenial)
exten = _[0-9A-Za-z].,n,Set(GROUP()= Potenial_${CONFNAME})
group_count(inbound)
group_count(inbound@conference)
group_count(potential@conference)
group_count(potential_${CONFNAME} )
however, when the call enters the conference, I need to unset the potential
calls groups
how do I do this ?
/me feels very very stupid
Julian
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your
machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
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Re: [asterisk-users] Speech recognition in asterisk using google voice API
this looks great - is there any chance of coverting the googletts.agi to use flac as well ? Julian On 4 January 2012 09:06, Lefteris Zafiris zaf@gmail.com wrote: On 01/04/2012 07:51 AM, Bruce B wrote: And with recent version 14.3.2 I get: /usr/local/bin/sox FAIL formats: no handler for file extension `flac' -- speech-recog.agi: /usr/local/bin/sox failed: 512 -- SIP/-002eAGI Script speech-recog.agi completed, returning 0 Regards, On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote: Very interesting. I just tried to get it to work but it complains about sox. Probably you used a different version of sox? *PBX-*CLI /usr/bin/sox: invalid option -- -* */usr/bin/sox: invalid option -- n* */usr/bin/sox: invalid option -- o* */usr/bin/sox: -r must be given a positive integer* * -- speech-recog.agi: /usr/bin/sox failed: 512* I am using: *Package sox-12.18.1-1.el5_5.1.i386 * Thanks, Note to self: Never release anything asterisk related without testing on RHEL/Centos 5 Thank you for reporting this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Speech recognition in asterisk using google voice API
the only reason is that I didn't want to have to install sox. Lazy. that's all ;) Just another piece of software to find and install running on amazon ec2, is the best thing to download the source and compile sox ? Thanks Julian On 4 January 2012 14:18, Lefteris Zafiris zaf@gmail.com wrote: On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote: this looks great - is there any chance of coverting the googletts.agi to use flac as well ? Julian In googletts.agi we get the voice data from google in mp3 and we convert it in a format that asterisk can read and playback (slin). If we store it in flac asterisk wont be able to read it natively and we would have to convert it each time we want to play it back to the user. In the speech recognition script we have to convert the voice data in flac before sending it to google because that's the accepted format. Is there some particular reason you want the googletts.agi data in flac? Lefteris Zafiris -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Speech recognition in asterisk using google voice API
nope :( On 4 January 2012 14:29, Lefteris Zafiris zaf@gmail.com wrote: On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote: the only reason is that I didn't want to have to install sox. Lazy. that's all ;) Just another piece of software to find and install running on amazon ec2, is the best thing to download the source and compile sox ? Thanks It should be on your distro repos already. Lefteris Zafiris -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] googleapps calendar
Hi Terry I managed to get it working eventually. I think that it may have been a problem with neon , as I downgraded to .25 from .29, removed all modules and make distclean, make install It started working at this point ! What would be really great would be 1) manager events for new / removed calendars 2) manager command to reload / refresh calendars 3) manager events for new / removed events 4) manager events for alarms Julian On 30 October 2011 02:31, Terry Wilson twil...@digium.com wrote: I am trying to get googleapps calendar integrated with my system. However, following all the instructions that I can find it still fails. this is my config file: [myGoogleCal] type=caldav url=https://www.google.com/calendar/dav/myemail/events/ user=myemail secret=mypassword refresh=15 timeframe=60 I just tried with: [calendar4] type = caldav url = https://www.google.com/calendar/dav/m...@mygoogleappsdomain.net/events/ user = m...@mygoogleappsdomain.net secret = mysneakypassword refresh = 15 timeframe = 60 and 'calendar show calendars' shows my calendar as free, and 'calendar show calendar calendar4' shows an upcoming event. I did have to commit a fix where if you don't have a channel set for notification, it would cause a crash. I just committed that fix a couple of seconds ago. So, everything looks to be working fine for me. when I start asterisk, and type calendar show calendars I get genesis2*CLI calendar show calendars Calendar Type Status -- myGoogleCal caldav free however, there are no events in myGoogleCal, and every 15 minutes I get the message Unknown response to CalDAV calendar pug, request REPORT to /calendar/myemail/events/: Could not read status line: connection was closed by server Sounds like a communication issue. Is there a proxy server required to access the outside? Perhaps libneon wasn't compiled with SSL support or something? You could verify that the url is reachable via a web browser (should download a .ics file) or via using a command-line tool on the Asterisk box like 'curl' to test the url, user, and password. -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] googleapps calendar
using asterisk-10 on CentOS
I am trying to get googleapps calendar integrated with my system.
However, following all the instructions that I can find it still
fails. this is my config file:
[myGoogleCal]
type=caldav
url=https://www.google.com/calendar/dav/myemail/events/
user=myemail
secret=mypassword
refresh=15
timeframe=60
when I start asterisk, and type calendar show calendars I get
genesis2*CLI calendar show calendars
Calendar Type Status
--
myGoogleCal caldav free
however, there are no events in myGoogleCal, and every 15 minutes I
get the message
Unknown response to CalDAV calendar pug, request REPORT to
/calendar/myemail/events/: Could not read status line: connection
was closed by server
Has anyone managed to get this running ?
Julian
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
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[asterisk-users] Sipp and asterisk 10
Something must have changed since asterisk 1.2 ... ;) I used to be able to run a simple sipp test using ./sipp -sn uac -d 2 -s sipp-client 127.0.0.1 -l 1 however, with asterisk 10 I am getting a call failure with the words 1319103341.109277: Aborting call on unexpected message for Call-Id '20-18295@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 407 Proxy Authentication Required What I am trying to figure out is how / why / what is different that now asterisk requires proxy authentication for sipp, when it didn't before. Thanks Julian -- Julian Lyndon-Smith IT Director, dot.r http://www.dotr.com I don’t care if it works on your machine! We are not shipping your machine!” Follow dot.r on http://twitter.com/DotRlimited -- _ -- 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] 10.0.0 better than 2.0.0?
it has been mentioned that 10 is of course 2 ... think not in base 10 On 22 July 2011 22:26, Matthew J. Roth mr...@imminc.com wrote: Kevin P. Fleming: The versions all go to ten. Look, right across the board, ten, ten, ten and... Asterisk Users: Oh, I see. And most open source projects upgrade to two? Kevin P. Fleming: Exactly. Asterisk Users: Does that mean it's better? Is it any better? Kevin P. Fleming: Well, it's eight better, isn't it? It's not two. You see, most blokes, you know, will be running at two. You're on two here, all the way up, all the way up, all the way up, you're on two on your software. Where can you go from there? Where? Asterisk Users: I don't know. Kevin P. Fleming: Nowhere. Exactly. What we do is, if we need that extra push over the cliff, you know what we do? Asterisk Users: Put it up to ten. Kevin P. Fleming: Ten. Exactly. Eight better. Asterisk Users: Why don't you just make two better and make two be the top number and make that a little better? Kevin P. Fleming: [pause] Asterisk goes to ten. -- Sorry, couldn't resist. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Asterisk 1.6.2.18, Cisco 79XX not registering
It was my problem ;) https://issues.asterisk.org/view.php?id=18951 fixed in svn On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote: On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com and see if there is anything similar to your issue. I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Good by asterisk 1.4? Please not.
Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ? If not, then I am stuck on my current version of 1.4, and will not be able to upgrade to either of those two versions, even for security fixes. Julian On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com wrote: On 11-04-19 09:28 AM, Kristijan Vrban wrote: @digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 Once a branch moves into security mode; no more bug fixes will be applied. If a security issue affects the 1.4 branch, a new release will be created containing only that fix. 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) We'd asked you to retest the issue against a supported branch (Asterisk 1.8), then triage the issue from there. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Good by asterisk 1.4? Please not.
The point I was trying to make was that *anyone* on 1.4 who uses ciscos will be forced to move to 1.6 or 1.8 if they want any security fixes applying , as the patches will go into 1.4 svn where the bug is present. IOW if you uses cisco's and 1.4 then that's the end of the line for you. No more patches *or* security fixes. People should be aware of that. But thanks for the helpful insight. Julian On 19 April 2011 16:18, William Stillwell will...@stillwellsoft.com wrote: Its not really had to install 1.6 or 1.8 on a test box, and see if a phone connects to it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, April 19, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good by asterisk 1.4? Please not. Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ? If not, then I am stuck on my current version of 1.4, and will not be able to upgrade to either of those two versions, even for security fixes. Julian On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com wrote: On 11-04-19 09:28 AM, Kristijan Vrban wrote: @digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 Once a branch moves into security mode; no more bug fixes will be applied. If a security issue affects the 1.4 branch, a new release will be created containing only that fix. 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) We'd asked you to retest the issue against a supported branch (Asterisk 1.8), then triage the issue from there. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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 -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Good by asterisk 1.4? Please not.
1.4 svn has a nasty bug in it at the moment. Would love to see that fixed ;) https://issues.asterisk.org/view.php?id=18951 Julian On 15 April 2011 14:22, Satish Patel satish...@hotmail.com wrote: You know we don't have choise. I had remembered when we shifted 1.2 to first release of 1.4 and we had many issue. Same thing right now I'm dealing with 1.8 things take time to stabilized. Good luck!! -- Sent from my iPhone On Apr 15, 2011, at 8:33 AM, Kristijan Vrban vrban.l...@googlemail.com wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- 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 -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Samsung smt-i3100
Anyone had any experience of using this phone with asterisk ? Trying to find out if I can provision it using tftp / http Thanks Julian -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Remote Unix Connection
I think I've seen this where I am trying to start another instance of asterisk using safe_asterisk, when I already have an instance running Julian On 16 October 2010 22:36, Dan Journo d...@keshercommunications.com wrote: Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Play a number of files to a caller
Hi Tilghman , thanks for the help. ControlPlayback can't be used with ExternalIVR, can it ? We use ControlPlayback in our current dialplan, what I am wanting (in concept) is to have a meetme/conference room where one of the parties is a caller, and the other party a file to be controlplaybacked by the caller ;) Best I have come up with so far is to start an attendant menu, get some curl data, if blank WaitExten() loop back to start, if not ControlPlayback the data. If I also background() some music file, will that play while the loop is running ? I suspect that it will start again from the beginning. Julian On 29 August 2010 18:17, Tilghman Lesher tles...@digium.com wrote: On Sunday 29 August 2010 03:32:07 Julian Lyndon-Smith wrote: Still can't figure out how to fastforward / rewind the current file being played. core show application ControlPlayback -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Play a number of files to a caller
Thanks Steve, Not sure how this would allow the caller to ff / rw the file currently being played - would that portion have to be written in the external program ? Are there any examples of how to use externalivr anywhere (I can't find on google) TIA Julian On 29 August 2010 01:29, Steve Edwards asterisk@sedwards.com wrote: On Sat, 28 Aug 2010, Julian Lyndon-Smith wrote: I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. Check out externalivr(). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Play a number of files to a caller
Thanks for that - I had already seen that particular page. Still can't figure out how to fastforward / rewind the current file being played. Julian 2010/8/29 Ondrej Škopek skopekond...@gmail.com: http://www.voip-info.org/wiki/view/Asterisk+cmd+ExternalIVR On Sun, Aug 29, 2010 at 8:33 AM, Julian Lyndon-Smith aster...@dotr.com wrote: Thanks Steve, Not sure how this would allow the caller to ff / rw the file currently being played - would that portion have to be written in the external program ? Are there any examples of how to use externalivr anywhere (I can't find on google) TIA Julian On 29 August 2010 01:29, Steve Edwards asterisk@sedwards.com wrote: On Sat, 28 Aug 2010, Julian Lyndon-Smith wrote: I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. Check out externalivr(). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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 -- -- Ondrej Škopek -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Play a number of files to a caller
I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the file, and then jump back to the original dialplan entry to start again. However, If I want to redirect, then this external event would need to know their channel. 2) I thought of a meetme / conference, but then they would not be able to control the playback of the file, right ? Anyone got any other thoughts ? TIA Julian -- Restore the Kingdom rightly With powers of control, Regain a bit more pride again In things New Labour stole Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Mobile answer machine cut off
Hey Matt, thanks for the response. I know it sounds impossible. Hell, I sound like a user :) But it *is* happening. And only on the cisco phones. We're trying to lab it up right now. What should I be looking for in the sip debug ? Julian On 25 August 2010 08:17, Matt Riddell li...@venturevoip.com wrote: On 22/08/10 10:38 AM, Julian Lyndon-Smith wrote: Crap, sorry, meant to add that we are running 1.4 svn head Julian On 21 August 2010 23:38, Julian Lyndon-Smithaster...@dotr.com wrote: We are having some strange issue where a call from asterisk dials a mobile number. If the number answers, we put the call through to an agent SIP phone. All works fine. If, however, the call goes straight through to the mobiles voicemail service *and* the agent phone is a Cisco 79xx, then the call is dropped (from the mobile end) about 1 second into the call. If the SIP phone is an Aastra9133i, then there is no problem. Has anyone seen anything like this ? Heh, seems impossible! Um, maybe the voicemail beep is the same tone as a * and * is used to disconnect a call or something? Try doing a SIP debug and see what turns up. Also make sure it's 100% repeatable :D -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Mobile answer machine cut off
We are having some strange issue where a call from asterisk dials a mobile number. If the number answers, we put the call through to an agent SIP phone. All works fine. If, however, the call goes straight through to the mobiles voicemail service *and* the agent phone is a Cisco 79xx, then the call is dropped (from the mobile end) about 1 second into the call. If the SIP phone is an Aastra9133i, then there is no problem. Has anyone seen anything like this ? Thanks Julian -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Mobile answer machine cut off
Crap, sorry, meant to add that we are running 1.4 svn head Julian On 21 August 2010 23:38, Julian Lyndon-Smith aster...@dotr.com wrote: We are having some strange issue where a call from asterisk dials a mobile number. If the number answers, we put the call through to an agent SIP phone. All works fine. If, however, the call goes straight through to the mobiles voicemail service *and* the agent phone is a Cisco 79xx, then the call is dropped (from the mobile end) about 1 second into the call. If the SIP phone is an Aastra9133i, then there is no problem. Has anyone seen anything like this ? Thanks Julian -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] NVidia component out
It may be a bit outside Myth, but it's even further outside from Asterisk :) Sorry, can't help. Julian On 22 August 2010 01:33, Michelle Dupuis mdup...@ocg.ca wrote: I realize this is getting a bit outside myth...but hopefully someone can offer some ideas... I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although the dual DVI outputs work great, the driver just won't detect anything connected to the component video connector. Is anyone out there successfully using the component video out on their Nvidia card with a recent driver? -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] 'System' application in asterisk
You could always use the CURL function directly in the dialplan
Julian
On 10 August 2010 08:36, Tino t...@sparksupport.com wrote:
Hi Steve, thanks for your interest in this matter.
I will explain my requirement here.
In my asterisk server before an agent doing manual dial is allowed a call,
asterisk will make an http request (to my crm, do not worry about this part
) and get back an OK or something else. … if it receives OK, it allows the
call, otherwise we just play an unauthorized call recording to the agent.
We make the http request using a wget | perl command and we want to
capture the output of the wget | perl command.
On Tue, Aug 10, 2010 at 12:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
Un-top-posting...
On Tue, 10 Aug 2010, Tino wrote:
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
On Mon, Aug 9, 2010 at 11:51 PM, Danny Nicholas da...@debsinc.com
wrote:
I think this answer is no. system only returns ${SYSTEMSTATUS} as
SUCCESS or FAILURE to tell you that the command finished or died. You could
however do a bash AGI that would set a variable with the result of what you
would have sent to system
On Tue, 10 Aug 2010, Tino wrote:
Sorry Dany, I am new to agi scripting. If you do not mind can you please
give me a sample script for this. That would be really helpful to me.
Unless the output from your system command is trivial, you should parse it
in the AGI and set channel variables as needed.
If you provide a bit more detail, you may get a more specific answer.
System() may not be the best approach.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Delay between answer and pickup ?
Anyone got a clue ? (he asks in desperation!) Julian On 9 July 2010 17:48, Julian Lyndon-Smith aster...@dotr.com wrote: We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing hello ? Hello ? and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of asterisk don't show this discrepancy. We are using a local channel to dial a landline through a sip provider. When the call is answered, the agent's phone is then dialled. the logs go something like this [Jul 9 13:29:26] VERBOSE[23396] logger.c: [Jul 9 13:29:26] -- SIP/provider-0001ed6e is making progress passing it to Local/somenum...@dialleroutbound-4c93,2 [Jul 9 13:29:44] VERBOSE[23396] logger.c: [Jul 9 13:29:44] -- SIP/provider-0001ed6e answered Local/01577864...@dialleroutbound-4c93,2 .. [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Executing [*00...@diallerconnected:2] Dial(Local/somenum...@dialleroutbound-4c93,1, SIP/*0086*|5|iA(autoanswer)) in new stack [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- SIP/*0086*-0001ed73 answered Local/somenum...@dialleroutbound-4c93,1 .. as you can see, the call is answered at 13:29:44 and the agent gets called (auto-answer phones) at 13:29:46, yes if you listen to the call recording, there is a 6 second gap between the person saying hello and the agent being connected. Is it possible that the call was answered 5 seconds *before* I get notification of the answer ? i.e. is the provider taking too long notifying me of the answer ? Julian -- _ -- 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] Not detecting hangup
That looks like the option that will help a lot. Thanks. On 8 July 2010 23:21, Steve Edwards asterisk@sedwards.com wrote: [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? On Thu, 8 Jul 2010, Danny Nicholas wrote: First thought is that you can put a timeout on your calls, but that is just a band-aid. Also not fixing the source of the problem, but rtpholdtimeout and rtptimeout may help. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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] Delay between answer and pickup ?
We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing hello ? Hello ? and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of asterisk don't show this discrepancy. We are using a local channel to dial a landline through a sip provider. When the call is answered, the agent's phone is then dialled. the logs go something like this [Jul 9 13:29:26] VERBOSE[23396] logger.c: [Jul 9 13:29:26] -- SIP/provider-0001ed6e is making progress passing it to Local/somenum...@dialleroutbound-4c93,2 [Jul 9 13:29:44] VERBOSE[23396] logger.c: [Jul 9 13:29:44] -- SIP/provider-0001ed6e answered Local/01577864...@dialleroutbound-4c93,2 .. [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Executing [*00...@diallerconnected:2] Dial(Local/somenum...@dialleroutbound-4c93,1, SIP/*0086*|5|iA(autoanswer)) in new stack [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- SIP/*0086*-0001ed73 answered Local/somenum...@dialleroutbound-4c93,1 .. as you can see, the call is answered at 13:29:44 and the agent gets called (auto-answer phones) at 13:29:46, yes if you listen to the call recording, there is a 6 second gap between the person saying hello and the agent being connected. Is it possible that the call was answered 5 seconds *before* I get notification of the answer ? i.e. is the provider taking too long notifying me of the answer ? Julian -- _ -- 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] Not detecting hangup
We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian -- _ -- 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] Forwarding inbound mobiles
We have a need for up to a dozen UK mobile numbers to be forwarded to a UK landline. I know that I can just forward them, but was wondering if anyone knew of any deals / contracts with a UK mobile operator that would lessen the cost. At the moment we are looking at going with Vodafone . Thanks Julian -- _ -- 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] Evaluating Asterisk
Ted, We've been using Asterisk in-house since 2005, with 100 people connected. We are a call center, making approx 3000 inbound / outbound calls per day 6 days a week. We have interfaces to 90 ISDN lines and SIP providers. We use MOH, voicemail, queues etc etc, and record every call. Each agent is managed in realtime by our own custom software which was written to link our database application to asterisk using sockets and jabber. It works really well. We have estimated the savings to the company in the region of 750,000 UK pounds since inception. Drop me a line if you want any further info. Julian On 19 April 2010 14:06, Ted Foote t...@abscollect.com wrote: I am thinking of moving from a traditional PBX to an asterisk box. Many of my leadership group are skeptical of asterisk. So I was hoping to find a call center that is currently using this technology that would not mind spending some time on a conference call to address some concerns that my team has. Thanks Ted Foote Allied Business Services, Inc. 616-741-0437 -- _ -- 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 -- _ -- 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] SIP equivalent of zap c option
At the moment, we have a feature where if someone's sip extension is called, we also make another call to their mobile. We use the c option in the zap dialstring so that the user has to press # after answering to confirm the call (this prevents things like the answermachine grabbing the call if the mobile is switched off). We are now looking to move towards a sip provider to take all of our ISDN calls, so instead of using zap / isdn to call the mobile, we will be routing the call over a SIP trunk. Is there any feature of SIP that we can use in order to duplicate this functionality (i.e. have to press # to confirm the call) Thanks Julian -- _ -- 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] chan_local and Originate
There was a bug reported on this, I think ... yes #16581
Fixed in
r244070 | tilghman | 2010-02-01 11:46:32 -0600 (Mon, 01 Feb 2010)
Julian
On 17 February 2010 15:00, James Northcott / Chief Systems
ja...@chiefsystems.ca wrote:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API action to originate a call:
Action: originate
Priority: 1
Context: trunk
Callerid: 100
Channel: Local/1...@callback/n
Exten: 123456789
Variable: USERFIELD=127.0.0.1|USEREXT=123456789
WaitTime: 30
This is intended to first call extension 100 in the callback context,
and then when that is answered, call 123456789 in the trunk context. I
have the following in the callback context:
exten = 100,1,Answer
exten = 100,2,Wait(2)
exten = 100,3,NoOP(${ANSWERED} ${USEREXT})
exten = 100,4,AGI(getChannelState.agi|${USEREXT})
exten = 100,5,GotoIf($[${EXISTS(${ANSWERED})}]?6:2)
exten = 100,6,Set(CDR(accountcode)=${USERFIELD})
exten = 100,7,Set(__OriginalCallerNum=c2c ${USEREXT})
exten = 100,8,Goto(handleq,s,new)
exten = 100,9,Hangup
The getChannelState AGI script just waits until the call to 123456789 is
answered before putting the caller into a queue.
The problem is that the second leg of the Originate, the call to
123456...@trunk, never happens. Even though the first action at
1...@local is to Answer, the Originate action doesn't see this, so I just
get the AGI calls every 2 seconds for 30 seconds, and then everything
hangs up.
This code did work in a previous version of Asterisk, but I am not 100%
sure it worked in 1.4.22 - it may have broken before then.
If I replace Local/1...@callback/n with my direct SIP channel, the
Originate works as expected.
Can anyone tell me if I am using the Local channel incorrectly here? Or
did something about the Local channel change in recent 1.4 versions? Is
there a better way to do what I'm trying to do?
Thanks,
James
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[asterisk-users] request for testing: MixMonitor Mute
I have uploaded a patch for 1.4 and trunk that allows you to mute either or both parts of a mixmonitor recording. I would appreciate it if someone apart from me could test it and let me know how you get on. Thanks! Julian https://issues.asterisk.org/view.php?id=16740 for PCI-DSS compliance we are not allowed to record a credit card number is a MixMonitor file. However, we must record all conversations I have added a new feature to audiohooks so that you can mute either read / write or both types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. MixMonitor now has two manager commands 1) manager show command MuteMixMonitor Action: MuteMixMonitor Synopsis: Mute a channel in MixMonitor Privilege: none Description: Mutes a Mixmonitor Channel. Variables: Channel: Channel to mute. Direction: Which part to mute. read|write|both (from channel|to channel|both channels). 2) manager show command unMuteMixMonitor Action: unMuteMixMonitor Synopsis: unMute a channel in MixMonitor Privilege: none Description: unMutes a Mixmonitor Channel. Variables: Channel: Channel to unmute. Direction: Which part to unmute. read|write|both (from channel|to channel|both channels). -- _ -- 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] Inserting white noise / music / sound file into mixmonitor
A week or so ago, I explained that we need to blank our call recording when some sensitive information like credit cards where being discussed. With the lists help, I managed to find the pause/ unpause monitor commands. That works great. However (there is always a however), what that now means is that the length of the call does not match the length of the call recording, so adding stuff like this happened at 11:04 into the call now is out by the length of time of the pause :( I was wondering if it was possible to replace the voice on either leg with a sound file or something, but only in mixmonitor, as we obviously need to hear the person talking in order to take the details. Julian -- _ -- 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] Call recordings and sensitive information
During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- 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] Call tagging
Something similar along the lines of a previous email - has anyone developed, or is using, something similar to this http://www.veritape.com/wp-content/uploads/2009/11/veritape-call-tagging-module-description.pdf Julian -- _ -- 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] Call recordings and sensitive information
Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can “pause” and “un-pause” monitoring. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian Lyndon-Smith *Sent:* Monday, January 25, 2010 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recordings and sensitive information
Oh, crap. the second I send, I realize I use features.conf, right ? ;) Is there any other way of getting this into the dialplan ? I would rather not have to have the users pressing a key, but for software to intercept the appropriate point and perform some AMI command Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can “pause” and “un-pause” monitoring. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian Lyndon-Smith *Sent:* Monday, January 25, 2010 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recordings and sensitive information
This is crazy. Something about writing to the list gives me ideas ;) What I am looking for is show manager command pausemonitor ;) Thanks anyway, all. Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Oh, crap. the second I send, I realize I use features.conf, right ? ;) Is there any other way of getting this into the dialplan ? I would rather not have to have the users pressing a key, but for software to intercept the appropriate point and perform some AMI command Julian 2010/1/25 Julian Lyndon-Smith aster...@dotr.com Yeah, was looking at this - my issue is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? Julian 2010/1/25 Danny Nicholas da...@debsinc.com Check this link http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor Depending on your release, you can “pause” and “un-pause” monitoring. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian Lyndon-Smith *Sent:* Monday, January 25, 2010 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call recordings and sensitive information During a telephone conversation with a customer, they sometimes give card details over the phone. under the pci-dss regulations we are not allowed to record the conversation where the details are being given. Is there a mute command or pause that can be sent to MixMonitor ? How has anyone else solved this issue ? Many thanks Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom vs Polycom
Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more managerial phones than the base phones which will be used for one line only. TIA Julian -- _ -- 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] GXV3140 and Xlite video
Has anyone managed to get these two phones to make a video call to each other ? If so, care to share how the hell you managed ? the GXV is at the latest firmware, and xlite the latest download Asterisk 1.4 trunk TIA Julian -- _ -- 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] GXV3140 and Xlite video
Urgh. That means my problem is probably beyond my control. The xlite shows the video from the gxv, but as soon as I hit the send video button xlite segfaults. I was hoping that there was a magical don't crap out on me setting in xlite that someone as found. Nuts. Thanks anyway. Julian 2010/1/14 SIP s...@arcdiv.com: Julian Lyndon-Smith wrote: Has anyone managed to get these two phones to make a video call to each other ? If so, care to share how the hell you managed ? the GXV is at the latest firmware, and xlite the latest download Asterisk 1.4 trunk TIA Julian Yes. Have done it often. Needed the firmware in the GVX that suppoerted H264 or H263.1 or whatever it was that Xlite 3 uses. Other than that, it was rather straight-forward. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Yealink vs Aastra
We have a couple dozen Aastra 9133i phones in use - no problems encountered, they worked well for us. However, these are now discontinued. Does anyone have any views on the new product line up , or the Yealink phones ? Julian ___ -- 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] USB ISDN30
I'm just curious to know if anyone is using a usb 2.0 / ISDN30 (specifically EuroISDN) device. We are looking to purchase another pci card, but was wondering if anyone has any horror / success stories to share regarding a usb device. TIA Julian ___ -- 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] Interesting problem with IP's
HaHa!. That is so funny, made me splurt my coffee over the keyboard. lol Julian 2009/12/9 Danny Nicholas da...@debsinc.com: Just a guess, but the connection probably went from full to half duplex. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, December 08, 2009 8:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Interesting problem with IP's Have a trunk 1.4 asterisk, running on centos on the lan at work. A long story, but we had the entire work network on a public address range (90.1.0.x), going to a firewall, then out to the net. At home (192.168.1.x network) I have a router that connects to the firewall via a vpn tunnel. All was great. My cisco 7960 (192.168.1.100) was able to register with the asterisk server on 90.1.0.76 - and there was no audio problems whatsoever. I also must stress that I had nat=no and no nat-specific flags set in asterisk. However,the day came where the techs decided that we should be on a private internal network, and moved all of the devices onto a 10.0.x.x internal network. Needless to say, it wasn't an easy task. Now, although my vpn is connected to the new network, and I can access all of the machine as I used to be able to, I now only have 1-way audio on my phone !! (I can hear, and it gets progressively worse,the other party cannot hear me) Why would this have changed ? Do I need to do nat stuff now ? and why ? Interesting. Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Interesting problem with IP's
Have a trunk 1.4 asterisk, running on centos on the lan at work. A long story, but we had the entire work network on a public address range (90.1.0.x), going to a firewall, then out to the net. At home (192.168.1.x network) I have a router that connects to the firewall via a vpn tunnel. All was great. My cisco 7960 (192.168.1.100) was able to register with the asterisk server on 90.1.0.76 - and there was no audio problems whatsoever. I also must stress that I had nat=no and no nat-specific flags set in asterisk. However,the day came where the techs decided that we should be on a private internal network, and moved all of the devices onto a 10.0.x.x internal network. Needless to say, it wasn't an easy task. Now, although my vpn is connected to the new network, and I can access all of the machine as I used to be able to, I now only have 1-way audio on my phone !! (I can hear, and it gets progressively worse,the other party cannot hear me) Why would this have changed ? Do I need to do nat stuff now ? and why ? Interesting. Julian ___ -- 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] Setting up skype
That's my point - SFA comes with a g729 licence, so why can't it transcode to the DAHDI channel ? Thanks also for the info. Very useful. Julian 2009/12/6 Roeften roef...@gmail.com: From what I understand your sip client can handle g729 whereas for DAHDI you need transcoding to a|ulaw. I am using it with no problems (have g729 licenses as well though). A bit off topic, I have found some extra configuration that is not really in the docs (or I could not find them): fullname=Your full name country=gr language=en city=City province=Province phone_home=+fullinternationalnumber phone_office=+fullinternationalnumber email=y...@email.com homepage=http://www.example.com avatar=/var/lib/asterisk/images/skype100x100.jpg Just a note the country code has to be lower case (i.e GR would not work). Panos On Sun, Dec 6, 2009 at 9:40 AM, Julian Lyndon-Smith aster...@dotr.com wrote: Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well ? If so, why do I have the problem ? And would this affect local channels as well ? Julian 2009/12/6 Kevin P. Fleming kpflem...@digium.com: Julian Lyndon-Smith wrote: external = ddi = dial(skype) and got a load of static with WARNING[15328]: channel.c:3098 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) on the console. Fired up a sip client, made the same call, and all was ok. Any clues ? The clues are in the documentation; SkypeIn and SkypeOut use G.729 for nearly all calls, so handling calls via those paths requires a G.729 transcoder on the system if the target of the call will not also be using G.729. This is why the Skype For Asterisk license includes licenses for Digium's G.729 software transcoder as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up skype
Aha. That was it. Thanks. I could not see that advice in the documentation. I may be blind, but it may be helpful to include it somewhere. Thanks again Julian 2009/12/6 Kevin P. Fleming kpflem...@digium.com: Julian Lyndon-Smith wrote: That's my point - SFA comes with a g729 licence, so why can't it transcode to the DAHDI channel ? It comes with a license, but does not include the transcoding functionality itself. You need to download and install the appropriate Digium codec_g729 module for your system to enable transcoding using that license. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up skype
As I have no friends and no life I thought that I would set up my asterisk server with Skype. 1) Paid the $, got the licence, built and installed 2) create a business skype account (called company foo) 3) created a member of the business called bar 4) updated the skype conf file 5) restarted asterisk = skype show settings Skype For Asterisk Settings: engine_directory: /tmp data_directory: /var/spool/asterisk/skype defaultuser: bar bind_address: 0.0.0.0 bind_port: 0 rtp_address: 127.0.0.1 https_proxy: https_proxy_user: https_proxy_password: socks5_proxy: socks5_proxy_user: socks5_proxy_password: disable_tcpauto: no disable_udp: no debug: no = skype show users Skype Users bar: Logged In 6) added a test to extensions.conf exten = 123650,1,Dial(Skype/b...@my.personal.skype) exten = 123650,2,Hangup() and get a Everyone is busy/congested at this time (1:0/0/1) [Dec 5 20:15:08] -- Executing [123...@isdnspan1:2] Hangup(Zap/1-1, ) in new stack My skype client can find bar, but it is offline, so I can't place calls either Anyone know what I am doing wrong ?? (1.4 source svn trunk) TIA Julian ___ -- 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] Setting up skype
well, aint that a bugger. Just looked at my contacts list on skype and the bloody thing is working ... wtf ? Another question: I just tried calling in like this external = ddi = dial(skype) and got a load of static with WARNING[15328]: channel.c:3098 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) on the console. Fired up a sip client, made the same call, and all was ok. Any clues ? 2009/12/5 Julian Lyndon-Smith aster...@dotr.com: As I have no friends and no life I thought that I would set up my asterisk server with Skype. 1) Paid the $, got the licence, built and installed 2) create a business skype account (called company foo) 3) created a member of the business called bar 4) updated the skype conf file 5) restarted asterisk = skype show settings Skype For Asterisk Settings: engine_directory: /tmp data_directory: /var/spool/asterisk/skype defaultuser: bar bind_address: 0.0.0.0 bind_port: 0 rtp_address: 127.0.0.1 https_proxy: https_proxy_user: https_proxy_password: socks5_proxy: socks5_proxy_user: socks5_proxy_password: disable_tcpauto: no disable_udp: no debug: no = skype show users Skype Users bar: Logged In 6) added a test to extensions.conf exten = 123650,1,Dial(Skype/b...@my.personal.skype) exten = 123650,2,Hangup() and get a Everyone is busy/congested at this time (1:0/0/1) [Dec 5 20:15:08] -- Executing [123...@isdnspan1:2] Hangup(Zap/1-1, ) in new stack My skype client can find bar, but it is offline, so I can't place calls either Anyone know what I am doing wrong ?? (1.4 source svn trunk) TIA Julian ___ -- 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] Setting up skype
Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well ? If so, why do I have the problem ? And would this affect local channels as well ? Julian 2009/12/6 Kevin P. Fleming kpflem...@digium.com: Julian Lyndon-Smith wrote: external = ddi = dial(skype) and got a load of static with WARNING[15328]: channel.c:3098 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) on the console. Fired up a sip client, made the same call, and all was ok. Any clues ? The clues are in the documentation; SkypeIn and SkypeOut use G.729 for nearly all calls, so handling calls via those paths requires a G.729 transcoder on the system if the target of the call will not also be using G.729. This is why the Skype For Asterisk license includes licenses for Digium's G.729 software transcoder as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many lines do you use.
Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... Julian ___ -- 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] Changing labels on Phones
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a hotdesk type system where anyone can log on to an extension - however what I would love to do is relabel the phone with the current owner when this logon happens. I know that I can change the sip.conf and phones tftp file, however this is a big problem with the Cisco's as they take *forever* (ok, maybe 2 / 3 minutes) to reboot (VLAN problem) 1) Has anyone actually solved this VLAN issue with the cisco ? 2) Is there any way of changing a label without rebooting the phone ? TIA Julian ___ -- 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] Cisco router
I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
Thanks for the info. I didn't have any model in mind, just wondering what was required. Thanks again, much appreciated. Julian 2009/10/14 Jonathan Thurman jthurma...@gmail.com: Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy app timeout
Using 1.4 svn, I want to implent the busy application. With the following dialplan: [inboundqueue] exten = _X.,1,Answer() exten = _X.,n,Goto(dropcall,1) ... exten = dropcall,1,Busy(10) exten = dropcall,n,hangup() If I call any number in the inboundqueue, I get the following: [Oct 1 12:06:44] -- Executing [444...@isdnspan1:1] Answer(Zap/1-1, ) in new stack [Oct 1 12:06:44] -- Executing [444...@inboundqueue:2] Goto(Zap/1-1, 1?dropcall|1) in new stack [Oct 1 12:06:44] -- Goto (inboundqueue,dropcall,1) [Oct 1 12:06:44] -- Executing [dropc...@inboundqueue:1] Busy(Zap/1-1, 10) in new stack [Oct 1 12:06:44] == Spawn extension (inboundqueue, dropcall, 1) exited non-zero on 'Zap/1-1' why does the busy not wait for 10 seconds before dropping the zap channel ? show application Busy foxtrot*CLI -= Info about application 'Busy' =- [Synopsis] Indicate the Busy condition [Description] Busy([timeout]): This application will indicate the busy condition to the calling channel. If the optional timeout is specified, the calling channel will be hung up after the specified number of seconds. Otherwise, this application will wait until the calling channel hangs up. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Static on the line randomly
We've been having a strange problem all day where when making outbound calls, all we get is static on the far end (i.e we can hear, they can't). We've restarted asterisk a couple of times to no avail. It now transpires that it is only mobile numbers that are affected (not all mobile networks, not all of the time) Is this a supplier problem (BT ISDN/32) ? Thanks Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap / dahdi errors
getting some errors on my test system. this is 1.4 (Asterisk SVN-branch-1.4-r214194) with a 4 port T412p card. Three of the ports are connected: Span 1 to the PSTN on a 10 channel bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4 is not plugged in. This has been working fine for several months. I updated a few days ago to the latest 1.4 branch. However, now I cannot dial into the system on span 1. I get a busy signal, and the following on the console: Is this a supplier issue, or my issue ?? TIA Julian [Aug 28 12:06:11] ERROR[10552]: chan_dahdi.c:8749 dahdi_pri_error: ACK received for '0' outside of window of '8' to '9', restarting [Aug 28 12:06:11] == Primary D-Channel on span 1 down [Aug 28 12:06:11] WARNING[10552]: chan_dahdi.c:2790 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Aug 28 12:06:11] == Primary D-Channel on span 1 up [Aug 28 12:06:11] ERROR[10552]: chan_dahdi.c:8749 dahdi_pri_error: !! Got a UA, but i'm in state 7 foxtrot*CLI zap show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3OK 0 0 0 T4XXP (PCI) Card 0 Span 4RED0 0 0 The 'zap show status' command is deprecated and will be removed in a future release. Please use 'dahdi show status' instead. [Aug 28 12:02:21] ERROR[10552]: chan_dahdi.c:8749 dahdi_pri_error: ACK received for '0' outside of window of '9' to '11', restarting [Aug 28 12:02:21] ERROR[10552]: chan_dahdi.c:8749 dahdi_pri_error: !! Got reject for frame 0, but we have nothing -- resetting! [Aug 28 12:02:21] ERROR[10552]: chan_dahdi.c:8749 dahdi_pri_error: !! Got a UA, but i'm in state 7 I have the following zapata.conf context=isdnspan1 pridialplan=unknown group=1 signalling=pri_cpe switchtype=euroisdn channel = 1-10 context=isdnspan2 pridialplan=unknown group=2,2 signalling=pri_net switchtype=euroisdn channel = 32-46,48-62 context=isdnspan3 pridialplan=unknown group=3,5 signalling=pri_cpe switchtype=euroisdn channel = 63-77,79-93 context=isdnspan4 pridialplan=unknown group=4,4 signalling=pri_cpe switchtype=euroisdn channel = 94-108,110-124 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] password length of sip peer
I'm trying to figure out the maximum length of a cisco 7960 password in the SIPmac.cfg file. An Aastra9133i can take at least a 36-character password, but the cisco craps out (can't authenticate) In order to stop me from doing a brute-force test, does anyone know the password lengths of Aastra 9133i Grandstream gxp2010 Cisco 7960/7940 ? Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk???
Nope - but you are also running on an unsupported version of asterisk, so I am not surprised. From the readme: ===[ Installation Overview ]=== It is required that the proper version of Asterisk is installed prior to installing Skype For Asterisk. Skype For Asterisk is currently supported on: Asterisk 1.4 versions = 1.4.25 Asterisk 1.6.0 versions = 1.6.0.6 Asterisk 1.6.1 versions = 1.6.1.5 Previous versions of Asterisk WILL NOT work properly with Skype For Asterisk. It is also important to make sure that the major version of Skype For Asterisk downloaded matches the version of Asterisk installed on the system. Trying to compile Skype For Asterisk 1.4 versions on Asterisk 1.6.0 while fail, etc. There is no version of Skype For Asterisk for Asterisk trunk. Julian 2009/8/19 Remco Barendse aster...@barendse.to: On Tue, 18 Aug 2009, Terry Wilson wrote: That does sound a bit pricey, although it it's as stable as the latest beta, I wont be buying it at all. Have you posted a bug describing the issues you are having at http://betareports.digium.com/mantis/ yet? I would love to have the opportunity to actually fix any bugs that people find. :-) I installed the 1.0 release of Skype for Asterisk and last night on my production box running Asterisk 1.26.1 i got segfaults and 32 core dumps, all happened in a time frame between 01:04 - 01:08 at night (so 4 minutes). Anyone else seeing this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Truecall
What is interesting is that there is no mention of the software used - if it is asterisk, he would need to make the code available, no ? Julian 2009/7/18 Alan Lord (News) alansli...@gmail.com: On 18/07/09 00:35, Gavin Henry wrote: This has to be an Asterisk based appliance no? http://www.truecall.co.uk/acatalog/trueCall_Features.html I saw this on the TV the other night. Couldn't believe how the dragons all thought it was such a cool idea. I was shouting at the telly saying You could do that with Asterisk very easily... Granted, if he's made the box, built it on an embedded SoC device then fair play, but he needs to have something Unique or anyone can do it. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
We use a queue so that we can have all the benefits of the queue whilst finding an agent : music on hold, periodic announcements etc etc. You are right - with a little more effort we could probably remove the need for the queue. But why would I do that if I can use the queue for the bits I want ;) Julian 2009/7/17 Alex Balashov abalas...@evaristesys.com: The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Um, I really don't know - we just use the periodic messages to play the traditional Your call is important to use (whatever the wording..) Julian. 2009/7/17 Alex Balashov abalas...@evaristesys.com: What value do the queue announcements (I am assuming these are pertaining to expected hold time, etc.) if there is only one agent? We use a queue so that we can have all the benefits of the queue whilst finding an agent : music on hold, periodic announcements etc etc. You are right - with a little more effort we could probably remove the need for the queue. But why would I do that if I can use the queue for the bits I want ;) Julian 2009/7/17 Alex Balashov abalas...@evaristesys.com: The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
Re: [asterisk-users] Skill based routing
Heh. See my previous posts ;) We use curl to grab the agent info from the application. Julian 2009/7/17 Leif Madsen leif.mad...@asteriskdocs.org: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? Can anyone share their ideas on this? I haven't built it yet, but have the idea of just using Local channels, placed in a queue, which when a call comes into the queue sets some channel variables (and making them transitive so they are available on the other side), then when the Queue calls the Local channel, to perform lookups from the set variables that verifies the call should be sent to the agent. If so, then it allows the call to go through and uses the Dial() in the Local channel to call the agent. Otherwise, it just hangs up, which then places the call back into the Queue, and will then just find a new agent. I'm sure there are a few other ways to do it, and there may be some disadvantages to my idea, but it seems pretty straight forward :) Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is Asterisk reliable for a call center application??
Um, yes ... Been using it for a call center since 2005. Julian 2009/7/12 Alex Balashov abalas...@evaristesys.com For 50 seats? I think so. gergis.rasmy wrote: i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers ___ -- 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 -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream 2010 and blinky lights
Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. 1) When the agent logs in / logs out, we rewrite the part of the dialplan for the hints and reload the dialplan 10 seconds after the *last* login / logout 2) For the MWI, we give each phone a fake voicemail (let's say _0001_). When an agent logs in, we link /var/spool/asterisk/voicemail/_0001_ to /var/spool/asterisk/voicemail/[mailbox] (where [mailbox] is the mailbox of the agent) and when they log out, we remove /var/spool/asterisk/voicemail/_0001_ This seems to work - the MWI lights up / off depending on the new vm within a couple of seconds 3) When checking for voicemail, each phone is configured to dial - the dialplan then checks the callerid (set by #1) and gets the mailbox for the agent. As I said, a bit of a hack, but it works for me ;) I know that this won't work for 1.6, but we are coming up with an alternative plan using Minivm Julian Andrew Thomas wrote: The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten = 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I used a mixture of php and mysql) - but even these are not ideal (especially if you have a large dial plan). Clue : exten 1234,hint,SIP/ABC works - you just need to change the ABC bit every time an agent logs in our out. This then gives you the lovely job of lighting any MWI lamps for that user as well. Oh the joys of Asterisk and hotdesking! HTH Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 02 July 2009 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream 2010 and blinky lights I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
Try client instead of component. Make sure that you selected the component in the menu select as well I can assure you that it works, and that it works well. We use it ;) Julian jonas kellens wrote: I have installed gnutls and gnutls-devel from RedHat repositories [r...@asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [r...@asterisk asterisk]# cd /usr/src/iksemel-1.3/ [r...@asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [r...@asterisk asterisk]# make [r...@asterisk asterisk]# make check [r...@asterisk asterisk]# make install [r...@asterisk asterisk]# ls -l /usr/lib | grep iksemel -rw-r--r-- 1 root root 184210 2009-07-06 14:52 libiksemel.a -rwxr-xr-x 1 root root 816 2009-07-06 14:52 libiksemel.la lrwxrwxrwx 1 root root 19 2009-07-06 14:52 libiksemel.so - libiksemel.so.3.1.0 lrwxrwxrwx 1 root root 19 2009-07-06 14:52 libiksemel.so.3 - libiksemel.so.3.1.0 -rwxr-xr-x 1 root root 138938 2009-07-06 14:52 libiksemel.so.3.1.0 Then compiled Asterisk again : [r...@asterisk asterisk]# cd /usr/src/asterisk-1.4.25.1/ [r...@asterisk asterisk]# make clean [r...@asterisk asterisk]# ./configure [r...@asterisk asterisk]# make menuconfig [r...@asterisk asterisk]# make [r...@asterisk asterisk]# make install Then edited jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=component ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not ;usesasl=yes;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100;;Timeout on the message stack. Then start Asterisk : [r...@asterisk asterisk]# /usr/sbin/asterisk -c And this is the error concerning jabber when wanting to connect to my OpenFire-server: [Jul 6 15:15:36] JABBER: reconnecting. [Jul 6 15:15:36] JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:component:accept' to='aster...@192.168.2.5' version='1.0' [Jul 6 15:15:36] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream from=openfire.jocan.local id=7pI2f xmlns=jabber:component:accept xmlns:stream=http://etherx.jabber.org/streams; version=1.0stream:error xmlns:stream=http://etherx.jabber.org/streams;bad-namespace-prefix xmlns=urn:ietf:params:xml:ns:xmpp-streams//stream:error [Jul 6 15:15:36] JABBER: asterisk OUTGOING: handshake2313234e99edf2891db7901990cf854e8e5639c3/handshake [Jul 6 15:15:36] JABBER: asterisk INCOMING: /stream:stream [Jul 6 15:15:40] WARNING[23732]: res_jabber.c:1573 aji_recv_loop: JABBER: socket read error [Jul 6 15:15:40] JABBER: reconnecting. [Jul 6 15:15:40] JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:component:accept' to='aster...@192.168.2.5' version='1.0' [Jul 6 15:15:40] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream from=openfire.jocan.local id=3oygw xmlns=jabber:component:accept xmlns:stream=http://etherx.jabber.org/streams; version=1.0stream:error xmlns:stream=http://etherx.jabber.org/streams;bad-namespace-prefix xmlns=urn:ietf:params:xml:ns:xmpp-streams//stream:error [Jul 6 15:15:40] JABBER: asterisk OUTGOING: handshakecccff622b0bafbf9db1e22034292e62610d93f48/handshake [Jul 6 15:15:40] JABBER: asterisk INCOMING: /stream:stream I don't know why connecting my Asterisk to my OpenFire (192.168.2.5) fails... Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
usetls=no Julian jonas kellens wrote: On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote: I can assure you that it works, and that it works well. We use it ;) My jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100;;Timeout on the message stack. Then I get the following : [Jul 6 20:07:57] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client from=openfire.jocan.local id=56ff9859 xml:lang=en version=1.0stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismDIGEST-MD5/mechanismmechanismPLAIN/mechanismmechanismANONYMOUS/mechanismmechanismCRAM-MD5/mechanism/mechanismscompression xmlns=http://jabber.org/features/compress;methodzlib/method/compressionauth xmlns=http://jabber.org/features/iq-auth/register xmlns=http://jabber.org/features/iq-register//stream:features [Jul 6 20:07:57] JABBER: asterisk OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='DIGEST-MD5'/ [Jul 6 20:07:57] JABBER: asterisk INCOMING: challenge xmlns=urn:ietf:params:xml:ns:xmpp-saslcmVhbG09Im9wZW5maXJlLmpvY2FuLmxvY2FsIixub25jZT0iSngyRVZCRmlDNlI4K1hlMU5rbm9PUUNWT1VEN1pGMEpXcnRydUxjdiIscW9wPSJhdXRoIixjaGFyc2V0PXV0Zi04LGFsZ29yaXRobT1tZDUtc2Vzcw==/challenge [Jul 6 20:07:57] JABBER: asterisk OUTGOING: response xmlns='urn:ietf:params:xml:ns:xmpp-sasl'dXNlcm5hbWU9ImFzdGVyaXNrIixyZWFsbT0ib3BlbmZpcmUuam9jYW4ubG9jYWwiLG5vbmNlPSJKeDJFVkJGaUM2UjgrWGUxTmtub09RQ1ZPVUQ3WkYwSldydHJ1TGN2Iixjbm9uY2U9IjQzZTVmYjFkNjZiMTU2OGI1MDFjNzk0ZDQ0MzMyYzFiIixuYz0wMDAwMDAwMSxxb3A9YXV0aCxkaWdlc3QtdXJpPSJ4bXBwLzE5Mi4xNjguMi41IixyZXNwb25zZT1kNGUxYzQ0ZDM0OGNjNWJkN2E2MzJiNzdmZjRjZTQ0OCxjaGFyc2V0PXV0Zi04/response [Jul 6 20:07:57] JABBER: asterisk INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslnot-authorized//failure [Jul 6 20:07:57] ERROR[24565]: res_jabber.c:606 aji_act_hook: JABBER: encryption failure. possible bad password. I am 100% sure I have the correct password ! I even took a very simple password without any special characters... Can you advise ?? Jonas. ___ -- 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] Grandstream 2010 and blinky lights
I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] using http to provision a Grandstrea GXP2000 phone
I have a GXP2010 phone, the one with 18 blinky lights ;) I currently provision the phone by writing out the conf file, encoding it and sending it to the tftp server. I was wondering if anyone had managed to automate the web side of provisioning ? TIA Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] hotdesk and voicemail
We have several types of phones, cisco 7940/7960 aastra 55i/9113i/ grandstream gxp2010 I want to be able to implement hotdesking where an agent will logon to any phone. I got all of that working, without having to reboot phones, but then hit a brick wall. Voicemail. I still want each phone to use the BLF for voicemail indication, and to use the voicemail button to dial voicemail directly. Is it possible to do this dynamically, or will I have to rewrite the phone config and reboot ? The issue I have with rebooting is that the cisco's take so bloody long to reboot (mainly waiting around at the VLAN) that it is unusable. Does anyone have any solutions to make the VLAN problem go away ? We don't use cisco switches. TIA Julian ___ -- 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] Can I run two instances of asterisk
Can I run two instances of asterisk sharing a single te412p ? I want to be able to have several asterisk servers (for testing various scenarios) running on one server. I was wondering if these asterisk processes could share a zaptel/dahdi card nicely. Julian ___ -- 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] Is there documentation explaining res_config_curl?
Eric Chamberlain wrote: On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: Eric Chamberlain wrote: [snip] Thank you, that bug does have useful information. We are working on moving from res_config_odbc to res_config_curl, so all asterisk requests go through our django backend, rather than django and asterisk sharing database tables. We had a buggy odbc driver (a 3rd party closed one) - we went from 2-3 crashes per day to zero in the last year, running nearly 3M config_curl requests per month now ;) It's, like, wow man ! Julian -- Eric Chamberlain, Founder RF.com - http://RF.com/ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] Is there documentation explaining res_config_curl?
Eric Chamberlain wrote: Is there any documentation that explains res_config_curl? We use the 1.4 backported version - it works so well I just can't sing it's praises enough. We use it for realtime voicemail and realtime queues / queue members. Have a look at bug #11747 for some documentation. Julian Specifically, the format of realtime calls made to the web server and what the return string for each call should look like? -- Eric Chamberlain ___ -- 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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] IVR Survey
See comments inline:
Steve Edwards wrote:
On Fri, 10 Apr 2009, James A. Shigley wrote:
Here is more or less what I'm trying to accomplish
1. Call comes in Plays Greeting
2. Starts Survey
3. Ask Q1, Record the answer (voice responses) repeat this step for
each Question
4. Combined the recorded responses into one file.
5. Email Combined Audio Fi
But I'm clueless as to how to combined the recordings into one file. I
don't want the questions in the recordings, Only the caller's side of
the conversation without the dead space while they listen to the
Qs/Think on their response.
sox *.wav combined.wav
Something I saw the other day which worked well for me: The a option
on the Record App (append to existing recording).
Julian
And since this isn't a vmail account and trying to avoid an AGI script
if possible I'm not sure how to email the recording(s).
You should embrace AGI. It is the solution to a large set of problems.
Here's a shell script snippet to email a binary file:
(
echo MIME-Version: 1.0
echo Content-ID: $(date)
echo Content-Transfer-Encoding: base64
echo Content-Type: application/octet-stream; name=\${FILE_NAME}\
echo Subject: ${SUBJECT}
${BASE64} ${FILE}
) | sudo /usr/sbin/sendmail -f ${FROM} ${TO}
I also want to be able to structure the body of the email so that it
reads something like
You have a new call from $CallerID - $CallerName on 'DateTime' ...
ect, ect.
Expand on the snippet above -- left as a exercise for the reader :)
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] h exten no getting run ...
Let me turn the question around slightly:
Are there any circumstances under which the h extension _won't_ get run ?
Julian
Julian Lyndon-Smith wrote:
Steve Edwards wrote:
On Sun, 29 Mar 2009, Julian Lyndon-Smith wrote:
Steve Edwards wrote:
Please show us the output from dialplan show questionnaire-menu.
Here you go
show dialplan questionnaire-menu
[ Context 'questionnaire-menu' created by 'pbx_config' ]
'0' =1. Goto(s|mainmenu)
[pbx_config]
'h' =1. Verbose(0|==
) [pbx_config]
's' =1. set(TIMEOUT(digit)=3)
[pbx_config]
2. set(TIMEOUT(response)=5)
[pbx_config]
[mainmenu] 3. Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
[pbx_config]
4. Goto(1|1)
I don't see an extension for 1.
Erm : In the previous email I sent there was
[pbx_config]
't' =1. Goto(s|mainmenu)
[pbx_config]
'_X' = 1. GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished|1|1) [pbx_config]
2. Set(QUESTION=${EXTEN})
[pbx_config]
3. Gosub(get-answer|Q|1)
[pbx_config]
[next] 4. Goto(${MATH(${EXTEN}+1|i)}|1)
[pbx_config]
Include ='questionnaire-hangup'
after the goto(1|1)
You may have missed it ;)
Thanks
Julian
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Solved : Re: h exten no getting run ...
I eventually found the problem - the h extension was getting run on the Zap channel as soon as the bridge between the SIP client and Zap client was broken. This is because of changes made to the cdr code in 1.4 trunk. However, the problem would not manifest itself to anyone except those using a backported version of app_queue.c that allows for the call to continue when the agent hangs up. For posterities sake, you need to add the following line of code to app_queue.c, in or around line 2784 @@ -2784,6 +2789,9 @@ *tries = qe-parent-membercount; *noption = 1; break; + case 'c': + ast_set_flag((bridge_config.features_caller), AST_FEATURE_NO_H_EXTEN); + break; case 'i': forwardsallowed = 0; break; Thanks for all the help and pointers - Steve, I'm getting to like templates ;) Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] h exten no getting run ...
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten is
not being run ?
console output:
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:1]
Set(Zap/1-1, TIMEOUT(digit)=3) in new stack
[Mar 29 10:33:49] -- Digit timeout set to 3
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:2]
Set(Zap/1-1, TIMEOUT(response)=5) in new stack
[Mar 29 10:33:49] -- Response timeout set to 5
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:3]
Playback(Zap/1-1,
custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0) in new stack
[Mar 29 10:33:49] -- Zap/1-1 Playing
'custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0' (language 'en')
[Mar 29 10:33:53] -- Channel 0/1, span 1 got hangup request, cause 16
[Mar 29 10:33:53] WARNING[18721]: file.c:738 ast_readaudio_callback:
Failed to write frame
[Mar 29 10:33:53] == Spawn extension (questionnaire-menu, s, 3) exited
non-zero on 'Zap/1-1'
[Mar 29 10:33:53] -- Hungup 'Zap/1-1'
[Mar 29 10:33:53] == End MixMonitor Recording Zap/1-1
===
[questionnaire-hangup]
exten =
h,1,Set(DATA=${CURL(MyApp/SaveQuestionnaire,COMPLETED=${COMPLETED}QUESTIONNAIRE_GUID=${QUESTIONNAIRE_GUID})})
exten = h,n,return
[questionnaire-menu]
exten = _X,1,GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished,1,1)
exten = _X,n,Set(QUESTION=${EXTEN})
exten = _X,n,Gosub(get-answer,Q,1)
exten = _X,n(next),Goto(${MATH(${EXTEN}+1,i)},1)
exten = s,1,set(TIMEOUT(digit)=3)
exten = s,n,set(TIMEOUT(response)=5)
exten = s,n(mainmenu),Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
exten = s,n,Goto(1,1)
exten = t,1,Goto(s,mainmenu)
include = questionnaire-hangup;
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Re: [asterisk-users] h exten no getting run ...
Meh. Has anyone got any clue ? I'm trying to test this tomorrow and it
is obviously not going to pass ;)
I've replaced the include = with a h,1,NoOp(here) and verified it with
a show dialplan
but that didn't work either
Julian
Julian Lyndon-Smith wrote:
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten is
not being run ?
console output:
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:1]
Set(Zap/1-1, TIMEOUT(digit)=3) in new stack
[Mar 29 10:33:49] -- Digit timeout set to 3
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:2]
Set(Zap/1-1, TIMEOUT(response)=5) in new stack
[Mar 29 10:33:49] -- Response timeout set to 5
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:3]
Playback(Zap/1-1,
custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0) in new stack
[Mar 29 10:33:49] -- Zap/1-1 Playing
'custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0' (language 'en')
[Mar 29 10:33:53] -- Channel 0/1, span 1 got hangup request, cause 16
[Mar 29 10:33:53] WARNING[18721]: file.c:738 ast_readaudio_callback:
Failed to write frame
[Mar 29 10:33:53] == Spawn extension (questionnaire-menu, s, 3) exited
non-zero on 'Zap/1-1'
[Mar 29 10:33:53] -- Hungup 'Zap/1-1'
[Mar 29 10:33:53] == End MixMonitor Recording Zap/1-1
===
[questionnaire-hangup]
exten =
h,1,Set(DATA=${CURL(MyApp/SaveQuestionnaire,COMPLETED=${COMPLETED}QUESTIONNAIRE_GUID=${QUESTIONNAIRE_GUID})})
exten = h,n,return
[questionnaire-menu]
exten = _X,1,GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished,1,1)
exten = _X,n,Set(QUESTION=${EXTEN})
exten = _X,n,Gosub(get-answer,Q,1)
exten = _X,n(next),Goto(${MATH(${EXTEN}+1,i)},1)
exten = s,1,set(TIMEOUT(digit)=3)
exten = s,n,set(TIMEOUT(response)=5)
exten = s,n(mainmenu),Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
exten = s,n,Goto(1,1)
exten = t,1,Goto(s,mainmenu)
include = questionnaire-hangup;
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Re: [asterisk-users] h exten no getting run ...
Steve Edwards wrote:
Untopposting...
Ouch. Sorry.
Julian Lyndon-Smith wrote:
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten
is not being run ?
This is not a dialplan snippet, this is the console output.
Yup, got it the wrong way around. Sorry.
console output:
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:1]
[snip]
===
[questionnaire-hangup]
exten =
h,1,Set(DATA=${CURL(MyApp/SaveQuestionnaire,COMPLETED=${COMPLETED}QUESTIONNAIRE_GUID=${QUESTIONNAIRE_GUID})})
exten = h,n,return
Return returns from a gosub. Where are you gosubbing to
questionnaire-hangup?
That's a spurious line from my experiments. I am no gosubbing from
anywhere to questionnaire-hangup. Please ignore that.
[questionnaire-menu]
exten = _X,1,GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished,1,1)
exten = _X,n,Set(QUESTION=${EXTEN})
exten = _X,n,Gosub(get-answer,Q,1)
exten = _X,n(next),Goto(${MATH(${EXTEN}+1,i)},1)
exten = s,1,set(TIMEOUT(digit)=3)
exten = s,n,set(TIMEOUT(response)=5)
exten = s,n(mainmenu),Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
exten = s,n,Goto(1,1)
exten = t,1,Goto(s,mainmenu)
include = questionnaire-hangup;
On Sun, 29 Mar 2009, Julian Lyndon-Smith wrote:
Meh. Has anyone got any clue ? I'm trying to test this tomorrow and it
is obviously not going to pass ;)
I've replaced the include = with a h,1,NoOp(here) and verified it with
a show dialplan
Please show us the output from dialplan show questionnaire-menu.
Here you go
show dialplan questionnaire-menu
[ Context 'questionnaire-menu' created by 'pbx_config' ]
'0' =1. Goto(s|mainmenu)
[pbx_config]
'h' =1. Verbose(0|==
) [pbx_config]
's' =1. set(TIMEOUT(digit)=3)
[pbx_config]
2. set(TIMEOUT(response)=5)
[pbx_config]
[mainmenu] 3. Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
[pbx_config]
4. Goto(1|1)
[pbx_config]
't' =1. Goto(s|mainmenu)
[pbx_config]
'_X' = 1. GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished|1|1) [pbx_config]
2. Set(QUESTION=${EXTEN})
[pbx_config]
3. Gosub(get-answer|Q|1)
[pbx_config]
[next] 4. Goto(${MATH(${EXTEN}+1|i)}|1)
[pbx_config]
Include ='questionnaire-hangup'
[pbx_config]
-= 5 extensions (11 priorities) in 1 context. =-
I added the hardcoded h to try and track things down. I never get the
ZZ :(
Also, cranking the verbosity up and showing the console output may help.
verbosity was 3. Will crank it up and send it on.
FWIW, I prefer using templates for little snippets that get included
often instead of includes. It makes it easier to debug the dialplan.
Thanks for the info - I'll check up on templates.
Thanks for the help thus far.
Julian
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] h exten no getting run ...
errotan wrote:
That h extension belongs to [questionnaire-hangup] it won't run unless the
call hangs up while in that context.
If you look in the snippet I provided, [questionnaire-hangup] is
included into [questionnaire-menu] - the hangup code may be used in
several contexts, so rather than duplicate code it is much better to
include it.
Julian
On Sunday 29 March 2009 11.42.29 Julian Lyndon-Smith wrote:
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten is
not being run ?
===
= console output:
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:1]
Set(Zap/1-1, TIMEOUT(digit)=3) in new stack
[Mar 29 10:33:49] -- Digit timeout set to 3
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:2]
Set(Zap/1-1, TIMEOUT(response)=5) in new stack
[Mar 29 10:33:49] -- Response timeout set to 5
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:3]
Playback(Zap/1-1,
custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0) in new stack
[Mar 29 10:33:49] -- Zap/1-1 Playing
'custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0' (language 'en')
[Mar 29 10:33:53] -- Channel 0/1, span 1 got hangup request, cause 16
[Mar 29 10:33:53] WARNING[18721]: file.c:738 ast_readaudio_callback:
Failed to write frame
[Mar 29 10:33:53] == Spawn extension (questionnaire-menu, s, 3) exited
non-zero on 'Zap/1-1'
[Mar 29 10:33:53] -- Hungup 'Zap/1-1'
[Mar 29 10:33:53] == End MixMonitor Recording Zap/1-1
===
[questionnaire-hangup]
exten =
h,1,Set(DATA=${CURL(MyApp/SaveQuestionnaire,COMPLETED=${COMPLETED}QUESTION
NAIRE_GUID=${QUESTIONNAIRE_GUID})}) exten = h,n,return
[questionnaire-menu]
exten = _X,1,GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished,1,1)
exten = _X,n,Set(QUESTION=${EXTEN})
exten = _X,n,Gosub(get-answer,Q,1)
exten = _X,n(next),Goto(${MATH(${EXTEN}+1,i)},1)
exten = s,1,set(TIMEOUT(digit)=3)
exten = s,n,set(TIMEOUT(response)=5)
exten = s,n(mainmenu),Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
exten = s,n,Goto(1,1)
exten = t,1,Goto(s,mainmenu)
include = questionnaire-hangup;
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Re: [asterisk-users] h exten no getting run ...
Steve Edwards wrote:
On Sun, 29 Mar 2009, Julian Lyndon-Smith wrote:
Steve Edwards wrote:
Please show us the output from dialplan show questionnaire-menu.
Here you go
show dialplan questionnaire-menu
[ Context 'questionnaire-menu' created by 'pbx_config' ]
'0' =1. Goto(s|mainmenu)
[pbx_config]
'h' =1. Verbose(0|==
) [pbx_config]
's' =1. set(TIMEOUT(digit)=3)
[pbx_config]
2. set(TIMEOUT(response)=5)
[pbx_config]
[mainmenu] 3. Playback(custom/Set1/${QUESTIONNAIRE_GUID}-0)
[pbx_config]
4. Goto(1|1)
I don't see an extension for 1.
Erm : In the previous email I sent there was
[pbx_config]
't' =1. Goto(s|mainmenu)
[pbx_config]
'_X' = 1. GotoIf($[${EXTEN}
${QUESTIONNAIRE_MAX}]?questionnaire-finished|1|1) [pbx_config]
2. Set(QUESTION=${EXTEN})
[pbx_config]
3. Gosub(get-answer|Q|1)
[pbx_config]
[next] 4. Goto(${MATH(${EXTEN}+1|i)}|1)
[pbx_config]
Include ='questionnaire-hangup'
after the goto(1|1)
You may have missed it ;)
Thanks
Julian
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Strange warning message
Can anyone give me any idea on where to start looking for this ? 1.4 svn (ish) It has appeared twice in the last hour on a system that gets numerous inbound calls to the same number TIA Julian [Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 2 ^ [Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] Dahdi Error
Got this in the log, with no calls active. Is it a problem with my isdn line, or * ? [Mar 15 11:36:18] ERROR[29161]: chan_dahdi.c:8735 dahdi_pri_error: ACK received for '0' outside of window of '39' to '40', restarting [Mar 15 11:36:18] == Primary D-Channel on span 1 down [Mar 15 11:36:18] WARNING[29161]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] UK ISDN-30 and ANI
Hi Andrew Andrew Thomas wrote: Please explain (in English) what you mean by ANI. http://www.tech-faq.com/ani-automatic-number-identification.shtml Julian Thanks -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith -- Sent: 12 March 2009 10:21 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: [asterisk-users] UK ISDN-30 and ANI -- -- Has anyone in the UK got ANI to work on an inbound call ? -- -- Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 -- -- Julian -- -- __ -- This email has been scanned by the MessageLabs Email Security System. -- For more information please visit http://www.messagelabs.com/email -- __ -- -- ___ -- -- 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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] UK ISDN-30 and ANI
David Quinton wrote: On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith aster...@dotr.com wrote: Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 AFAIK (and our E1 doesn't go to * box) a) you mean CLI a) No I don't. CLI is different to ANI b) you have to pay BT extra for Calling Line Identity Presentation GBP7.50 / qtr on our last bill See a). We already have CLI. I need ANI ;) HTH ___ -- 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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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
