[asterisk-users] how send calls to gatekeeper?
hello everyone
i have a simple question: i have an asterisk which is a h323 gateway
and has a h323 connection to a cisco gatekeeper and a sip connection
to a pbx.
my question is: how can i send all calls to gatekeeper?
i searched a lot and found that i should set gatekeeper=192.168.0.X
(ip address of my gatekeeper) in h323.conf file.
but what about extensions.conf file? should i define an extension like
a simple h323 connection to gatekeeper (like
"exten=>_2.,1,Dial(H323/${EXTEN}@cisco_out,60,)")? or no dial pattern
need to be defined in extension.conf file? if we should define dial
pattern, what is different between a simple trunk h323 connection and
a gateway-gatekeeper h323 connection?
thanks in advance
SAM
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[asterisk-users] announcement to be played for attended transfer call
Hello List, I want to play an announcement for attended transfer calls. For example, "A" calls "B", "B" answers the call and transfers (attended) to "C" - once transfer is complete "B" should hear an announcement saying "you call has been transferred". Is there any configuration in asterisk to implement this behavior? I have not used asterisk Transfer Dialplan application or feature.conf for configuring the transfer; however I am using SIP REFER from UA to request the transfer. Regards, Rajib -- _ -- 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] Executing Script after MixMonitor is called
And yes if you want to use System application in your dialplan then have
System in your h extension
System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav
/PathToMp3FileToBE Stored/filename.mp3)
On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot wrote:
> Hi Gopamkrishnan,
>
> Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
> facility to execute a command when recording is over.
>
> *In my case, 'wav2mp3' is a script which gets executed and converts recorded
> wav audio file to mp3. I pass ${FILENAME} as an argument to my script.
> *
>
> *You should have something like
> *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) in
> your dialplan.
>
> Hope this helps.
>
> --Satish Barot
>
> Ahmedabad, India
>
>
>
>
>
> On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi Satish,
>>
>> I tried with sox, without any parameter, just sox filename.wav to
>> filename.mp3, in linux shell prompt... the file is been converted...
>>
>> Now If i want to run that command using dialplan,
>>
>> MixMonitor(filename.wav,m)
>> Monitor_Exec(sox filename.wav filename.mp3)
>>
>> Or to use System command?
>>
>> Regards..
>>
>>
>> On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot > > wrote:
>>
>>> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
>>> (Asterisk 1.8.7.0).
>>> ...
>>> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
>>> ^{FILENAME})
>>> ...
>>> and my script is...
>>>
>>> #!/bin/bash
>>>
>>> WAV="/var/spool/asterisk/monitor/$1"
>>> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
>>> MP3DEST="/var/spool/asterisk/mp3/$MP3"
>>> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m
>>> --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>>>
>>> --SATISH BAROT
>>> Ahmedabad,India.
>>>
>>>
>>> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib wrote:
>>>
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor
command is called but it doesnt execute the command, I tried the command in
terminal it worked, any help please ... below is my dial plan
exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8
-t -F -m m --bitwidth 8 --quiet
"/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
"/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
"/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
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>>>
>>>
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>>>
>>
>>
>> --
>> _
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>>http://www.asterisk.org/hello
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>>
>
>
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Re: [asterisk-users] Executing Script after MixMonitor is called
Hi Gopamkrishnan,
Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
facility to execute a command when recording is over.
*In my case, 'wav2mp3' is a script which gets executed and converts
recorded wav audio file to mp3. I pass ${FILENAME} as an argument to
my script.
*
*You should have something like
*MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav)
in your dialplan.
Hope this helps.
--Satish Barot
Ahmedabad, India
On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> Hi Satish,
>
> I tried with sox, without any parameter, just sox filename.wav to
> filename.mp3, in linux shell prompt... the file is been converted...
>
> Now If i want to run that command using dialplan,
>
> MixMonitor(filename.wav,m)
> Monitor_Exec(sox filename.wav filename.mp3)
>
> Or to use System command?
>
> Regards..
>
>
> On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot
> wrote:
>
>> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
>> (Asterisk 1.8.7.0).
>> ...
>> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
>> ^{FILENAME})
>> ...
>> and my script is...
>>
>> #!/bin/bash
>>
>> WAV="/var/spool/asterisk/monitor/$1"
>> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
>> MP3DEST="/var/spool/asterisk/mp3/$MP3"
>> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m --bitwidth
>> 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>>
>> --SATISH BAROT
>> Ahmedabad,India.
>>
>>
>> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib wrote:
>>
>>> Hello Guys,
>>> I am trying to convert files that are .wac to mp3 after mixmonitor
>>> command is called but it doesnt execute the command, I tried the command in
>>> terminal it worked, any help please ... below is my dial plan
>>> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8
>>> -t -F -m m --bitwidth 8 --quiet
>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
>>> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
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Re: [asterisk-users] Executing Script after MixMonitor is called
Hi Satish,
I tried with sox, without any parameter, just sox filename.wav to
filename.mp3, in linux shell prompt... the file is been converted...
Now If i want to run that command using dialplan,
MixMonitor(filename.wav,m)
Monitor_Exec(sox filename.wav filename.mp3)
Or to use System command?
Regards..
On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot wrote:
> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
> (Asterisk 1.8.7.0).
> ...
> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
> ^{FILENAME})
> ...
> and my script is...
>
> #!/bin/bash
>
> WAV="/var/spool/asterisk/monitor/$1"
> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
> MP3DEST="/var/spool/asterisk/mp3/$MP3"
> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m --bitwidth
> 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>
> --SATISH BAROT
> Ahmedabad,India.
>
>
> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib wrote:
>
>> Hello Guys,
>> I am trying to convert files that are .wac to mp3 after mixmonitor
>> command is called but it doesnt execute the command, I tried the command in
>> terminal it worked, any help please ... below is my dial plan
>> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8
>> -t -F -m m --bitwidth 8 --quiet
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
>> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>>
>> --
>> _
>> -- 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
>>
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
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>http://www.asterisk.org/hello
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>
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Re: [asterisk-users] Pulse Audio "Motorboating" Audio with Asterisk
Pulse Audio 4.0 just came out and has gotten good reviews as it improves
audio quality...I installed it on the devel and support mediaports and will
test tomorrow.
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/4.0/
On Mon, Jun 10, 2013 at 7:59 PM, Robert Krakora <
rob.krak...@messagenetsystems.com> wrote:
> https://bbs.archlinux.org/viewtopic.php?pid=920549
>
>
> On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis wrote:
>
>> When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
>> I get a motorboating sound or warble - or - just not clear audio.
>>
>> When I switch that to ALSA direct it sounds just fine.
>>
>> What might be happening with pulse audio that it does not
>> sound clear???
>>
>> asound.conf below.
>>
>> Thanks,
>>
>> Jerry
>>
>> more /etc/asound.conf
>> #
>> # Place your global alsa-lib configuration here...
>> #
>>
>> @hooks [
>> {
>> func load
>> files [
>> "/etc/alsa/pulse-default.conf"
>> ]
>> errors false
>> }
>> ]
>>
>>
>>
>>
>> --
>> __**__**_
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
>
> --
> Rob Krakora
> MessageNet Systems
> 101 East Carmel Dr. Suite 105
> Carmel, IN 46032
> (317)566-1677 Ext 212
> (317)663-0808 Fax
>
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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Re: [asterisk-users] Pulse Audio "Motorboating" Audio with Asterisk
https://bbs.archlinux.org/viewtopic.php?pid=920549
On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis wrote:
> When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
> I get a motorboating sound or warble - or - just not clear audio.
>
> When I switch that to ALSA direct it sounds just fine.
>
> What might be happening with pulse audio that it does not
> sound clear???
>
> asound.conf below.
>
> Thanks,
>
> Jerry
>
> more /etc/asound.conf
> #
> # Place your global alsa-lib configuration here...
> #
>
> @hooks [
> {
> func load
> files [
> "/etc/alsa/pulse-default.conf"
> ]
> errors false
> }
> ]
>
>
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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Re: [asterisk-users] no silk translation ?
On 06/10/2013 05:24 PM, Sean Darcy wrote: Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551...@voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make SIP/ng- compatible with Motif/+12025551...@voice.google.com-da3c == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-' core show translations doesn't include any SILK. SILK is installed: core show codec 100018 100018 SILK Custom Format 8khz 100018 SILK Custom Format 12khz 100018 SILK Custom Format 16khz 100018 SILK Custom Format 24khz sean Maybe it's because SILK is "unknown" : == Registered translator 'silktolin' from format unknown to slin, table cost, 90, computational cost 99 == Registered translator 'lintosilk' from format slin to unknown, table cost, 60, computational cost 99 == Registered translator 'silktolin12' from format unknown to slin12, table cost, 93, computational cost 99 == Registered translator '12lintosilk' from format slin12 to unknown, table cost, 875000, computational cost 99 == Registered translator 'silktolin16' from format unknown to slin16, table cost, 93, computational cost 99 == Registered translator '16lintosilk' from format slin16 to unknown, table cost, 875000, computational cost 99 == Registered translator 'silktolin24' from format unknown to slin24, table cost, 93, computational cost 99 == Registered translator '24lintosilk' from format slin24 to unknown, table cost, 875000, computational cost 99 codec_silk.so => (Silk Transcoder) sean -- _ -- 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] no silk translation ?
On 06/10/2013 05:24 PM, Sean Darcy wrote: Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551...@voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make SIP/ng- compatible with Motif/+12025551...@voice.google.com-da3c == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-' core show translations doesn't include any SILK. SILK is installed: core show codec 100018 100018 SILK Custom Format 8khz 100018 SILK Custom Format 12khz 100018 SILK Custom Format 16khz 100018 SILK Custom Format 24khz sean Maybe the reason SILK is not showing up, is because it's "unknown": == Registered translator 'silktolin' from format unknown to slin, table cost, 90, computational cost 99 == Registered translator 'lintosilk' from format slin to unknown, table cost, 60, computational cost 99 == Registered translator 'silktolin12' from format unknown to slin12, table cost, 93, computational cost 99 == Registered translator '12lintosilk' from format slin12 to unknown, table cost, 875000, computational cost 99 == Registered translator 'silktolin16' from format unknown to slin16, table cost, 93, computational cost 99 == Registered translator '16lintosilk' from format slin16 to unknown, table cost, 875000, computational cost 99 == Registered translator 'silktolin24' from format unknown to slin24, table cost, 93, computational cost 99 == Registered translator '24lintosilk' from format slin24 to unknown, table cost, 875000, computational cost 99 codec_silk.so => (Silk Transcoder) sean -- _ -- 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] "+" dialplan
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to double zeros if your provider can't handle it
; normal 00 prefix
exten => _00ZZXXX.,1,Macro(beforealldials)
exten => _00ZZXXX.,n,Dial(SIP/${EXTEN}@${OUTGOING_LINE})
exten => _00ZZXXX.,n,Hangup()
; swap + prefix to 00
exten => _+ZZXXX.,1,Macro(beforealldials)
exten => _+ZZXXX.,n,Dial(SIP/00${EXTEN:1}@${OUTGOING_LINE})
exten => _+ZZXXX.,n,Hangup()
regards
adam
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[asterisk-users] "+" dialplan
Hello guys, I looking for some dial plan which can mach on +xxx numbers instead of 00xxx numbers. Some users of main use + instead of 00 for international dial. Is there any solution for this problem? As far as i readed in asterisk is some kind of replacement of characters in dial plan command. Could i use that for archiving this option? Thank you for help. Jonson. -- _ -- 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] no silk translation ?
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551...@voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make SIP/ng- compatible with Motif/+12025551...@voice.google.com-da3c == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-' core show translations doesn't include any SILK. SILK is installed: core show codec 100018 100018 SILK Custom Format 8khz 100018 SILK Custom Format 12khz 100018 SILK Custom Format 16khz 100018 SILK Custom Format 24khz sean -- _ -- 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] DTLSv1_method on NetBSD
This is the second issue I found while trying to install Asterisk on a NetBSD box. I can't load the rtp module because HAVE_OPENSSL_SRTP seems to be set. Is there some way to simply force this variab;e to be unset from a configuration variable? -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net Voip: sip:da...@vex.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-users
[asterisk-users] Where is HAVE_NEWLOCALE set?
I am trying to build Asterisk on a NetBSD system but I am running into two problems. The first only happens on an installation built from NetBSD HEAD. The config variable HAVE_NEWLOCALE is erroneously set during configure but this system does not have newlocale(). I can't seem to find where this gets set to true. Interestingly a stable release of NetBSD does not have this issue although it still has the second issue which I will start a separate thread for. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net Voip: sip:da...@vex.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-users
Re: [asterisk-users] DAHDI-linux 2.7 compile error with CONFIG_DAHDI_NET enabled
On 06/10/2013 11:53 AM, Shaun Ruffell wrote: On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote: Not sure how I should officially report this... You should feel free to open issues at http://issues.asterisk.org. but I'm getting a compile error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h. I am able to compile successfully when I leave it undefined, but I need to be able to use the network support. /oct6100_api/oct6100_tsst.o AR /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/oct612x/lib.a Building modules, stage 2. MODPOST 0 modules make[1]: Leaving directory `/usr/src/linux-3.4.45' make -C /lib/modules/3.4.45-smp/build SUBDIRS=/tmp/dahdi-linux-2.7.0-net/drivers/dahdi DAHDI_INCLUDE=/tmp/dahdi-linux-2.7.0-net/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-3.4.45' CC [M] /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c: In function 'dahdi_net_open': /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c:1967:4: error: 'struct dahdi_chan' has no member named 'rxbufpolicy' make[2]: *** [/tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o] Error 1 make[1]: *** [_module_/tmp/dahdi-linux-2.7.0-net/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-3.4.45' make: *** [modules] Error 2 Thanks for reporting this. I have a patch [1] for the next release. If you are willing, care to apply it to your 2.7.0 tree and check it out? If you are building from a tarball you can easily apply it like: $ curl "http://git.asterisk.org/gitweb/?p=team/sruffell/dahdi-linux.git;a=patch;h=e4d89ffa7485"; | patch -p1 [1] http://git.asterisk.org/gitweb/?p=team/sruffell/dahdi-linux.git;a=commitdiff;h=e4d89ffa7485 Cheers, Shaun Thank you Shaun, that patch did the trick. DAHDI compiled and appears to be functioning normally. I wondered if I might impose upon you for a question. I am in the process of replacing an old router with a T1 interface with a Linux machine. My test rig is currently using a spare TE220F. I know digium's card were primarily designed to function in a telephony role, but is there any technical reason I should not use them in an exclusively data role as well? I am trying to decide if I should purchase another TE220F (which I have experience with) or use a Sangoma product (which I do not). Thank you for your time. -- Dave -- _ -- 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] Cut offs on outgoing SIP calls
Hey Philipp, I will try soon the new version and let you know. Currently my users are pointing to a PBX in my local-private network with no problems. When I use wireshark I see my internal peers trying to send the ACK packets 4 or 5 times until hangup, at the same time the PBX are requesting that very packet many times until it decides to hangup (as you can see in previous message). The funny thing happens when I restart my router, everything works fine, but 2 or 3 hours later calls start getting cut-offs again. I'm not very used to routers but if someone have some tip on Cisco 2811 it will be great. Definitely it's a NAT issue, any help is welcome. Elder D. Arohuanca Lima - Peru On Sat, May 18, 2013 at 8:10 PM, Philipp von Klitzing wrote: > Hi! > > > I've suffering cut offs after 6 or 7 seconds a call is answered, > > incoming calls are working fine, but outgoing ones show the gollowing > > messages when are being dropped > > [...] > > It seems the SIP ACK is not being received properly. > > I can confirm this issue: In my case it happens with calls coming in from > a patton ISDN gateway to Asterisk 1.8.20.1. > > The calls is processed and passed to a snom phone, audio flows fine for a > few seconds, but then Asterisk terminates the call. Interestingly this > never happens on internal calls (from snom to snom). Downgrading to > Asterisk 1.4 makes the issue go away as well. > > Have you tried 1.8.22? I haven't yet, but it seems to come with a fix for > a deadlock in the SIP channel which *might* solve the issue we are both > experiencing (see ASTERISK-21389). > > Philipp > > > -- > _ > -- 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
Re: [asterisk-users] DAHDI-linux 2.7 compile error with CONFIG_DAHDI_NET enabled
On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote: > > Not sure how I should officially report this... You should feel free to open issues at http://issues.asterisk.org. > but I'm getting a compile error with DAHDI-linux 2.7 when I define > CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h. I am able to > compile successfully when I leave it undefined, but I need to be > able to use the network support. > > > /oct6100_api/oct6100_tsst.o > AR /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/oct612x/lib.a > Building modules, stage 2. > MODPOST 0 modules > make[1]: Leaving directory `/usr/src/linux-3.4.45' > make -C /lib/modules/3.4.45-smp/build > SUBDIRS=/tmp/dahdi-linux-2.7.0-net/drivers/dahdi > DAHDI_INCLUDE=/tmp/dahdi-linux-2.7.0-net/include > DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules > DAHDI_BUILD_ALL=m > make[1]: Entering directory `/usr/src/linux-3.4.45' > CC [M] /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o > /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c: In function > 'dahdi_net_open': > /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c:1967:4: error: > 'struct dahdi_chan' has no member named 'rxbufpolicy' > make[2]: *** [/tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o] Error 1 > make[1]: *** [_module_/tmp/dahdi-linux-2.7.0-net/drivers/dahdi] Error 2 > make[1]: Leaving directory `/usr/src/linux-3.4.45' > make: *** [modules] Error 2 Thanks for reporting this. I have a patch [1] for the next release. If you are willing, care to apply it to your 2.7.0 tree and check it out? If you are building from a tarball you can easily apply it like: $ curl "http://git.asterisk.org/gitweb/?p=team/sruffell/dahdi-linux.git;a=patch;h=e4d89ffa7485"; | patch -p1 [1] http://git.asterisk.org/gitweb/?p=team/sruffell/dahdi-linux.git;a=commitdiff;h=e4d89ffa7485 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI-linux 2.7 compile error with CONFIG_DAHDI_NET enabled
Not sure how I should officially report this, but I'm getting a compile error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h. I am able to compile successfully when I leave it undefined, but I need to be able to use the network support. /oct6100_api/oct6100_tsst.o AR /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/oct612x/lib.a Building modules, stage 2. MODPOST 0 modules make[1]: Leaving directory `/usr/src/linux-3.4.45' make -C /lib/modules/3.4.45-smp/build SUBDIRS=/tmp/dahdi-linux-2.7.0-net/drivers/dahdi DAHDI_INCLUDE=/tmp/dahdi-linux-2.7.0-net/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-3.4.45' CC [M] /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c: In function 'dahdi_net_open': /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c:1967:4: error: 'struct dahdi_chan' has no member named 'rxbufpolicy' make[2]: *** [/tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o] Error 1 make[1]: *** [_module_/tmp/dahdi-linux-2.7.0-net/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-3.4.45' make: *** [modules] Error 2 -Dave -- _ -- 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] OC3/STM-1 Line Card
Adtran MX2800 is rock solid. Save some money and use NFAS. Thanks, Steve Totaro On Sun, Jun 9, 2013 at 10:11 PM, Nick Khamis wrote: > Thank you so much for your responses!!! With this route we would have > to manage so many * boxes with T1s, not to mention, the hit we would > take on the MUX. Any decent DS/T3 cards out there? > > N. > > -- > _ > -- 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
Re: [asterisk-users] Problem with dahdi XPP driver?
On Thu, Jun 6, 2013 at 10:12 AM, Matteo wrote: > > > > On Thu, Jun 6, 2013 at 9:56 AM, Tzafrir Cohen wrote: > >> On Thu, Jun 06, 2013 at 09:31:39AM +0200, Matteo wrote: >> > Hi list, >> > I had a problem with the dahdi XPP driver. >> > After this error in syslog, the Xorcom disconnect from the server: >> >> Has it happened once? More then once? Reproducable? >> >> How long has the Astribank been working till then? >> > > Hi Tzafrir, > I don't know how many times it happened, this time I was reported because > they have had problems with HA. > This Astribank has been in production for more than 1 year, some months > ago (maybe in September) I have upgraded the dahdi version to 2.6.1 from an > older one (2.3). > > Hi Tzafir, any hint about this type of issue? Thanks, Matteo > > >> > (usb-:00:1d.7-3) [X1047686]: nonzero write bulk status received: >> -108 >> >> Is X1047686 the serial number of the Astribank? See 'dahdi_hardware -v' >> > > Yes it is the serial number of the Astribank. > > Regards, > Matteo > >> >> > >> > >> > Jun 3 15:03:29 kernel: [361010.637858] *NOTICE-xpp_usb: xusb-0 >> > (usb-:00:1d.7-3) [X1047686]: Sluggish USB. Dropping next PCM frame >> (p** >> > **ending_writes=5)* >> > Jun 3 15:03:52 kernel: [361033.890575]* ERR-xpp: XBUS-00: Failed to >> send >> > from command_queue (ret=-19)* >> > Jun 3 15:03:52 kernel: [361033.894565] [ cut here >> > ] >> > Jun 3 15:03:52 kernel: [361033.894565] WARNING: at >> kernel/softirq.c:141 >> > local_bh_enable+0x2f/0x6a() >> > Jun 3 15:03:52 kernel: [361033.894565] Hardware name: >> > Jun 3 15:03:52 kernel: [361033.894565] Modules linked in: >> > dahdi_echocan_oslec echo xpd_pri xpp_usb xpp dahdi crc_ccitt drbd cn >> ipv6 >> > loop rng_core serio_raw i2c_i801 ehci_hcd uhci_hcd iTCO_wdt i2c_core >> usbcore >> > Jun 3 15:03:52 kernel: [361033.894565] Pid: 0, comm: swapper Not >> tainted >> > 2.6.30.9 #3 >> > Jun 3 15:03:52 kernel: [361033.894565] Call Trace: >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > warn_slowpath_common+0x5e/0x8a >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > warn_slowpath_null+0xa/0xc >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > local_bh_enable+0x2f/0x6a >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> sk_filter+0x63/0x6c >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > netlink_broadcast+0x1aa/0x2e7 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > kobject_uevent_env+0x295/0x340 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > xbus_setstate+0x155/0x18d [xpp] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > xbus_command_queue_tick+0x15d/0x18c [xpp] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > xframe_receive_pcm+0x91/0xe28 [xpp] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > getnstimeofday+0x4d/0xca >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > getnstimeofday+0x4d/0xca >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > xframe_receive+0x118/0x52c [xpp] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > do_gettimeofday+0xf/0x29 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > xpp_receive_callback+0x117/0x13e [xpp_usb] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > usb_hcd_giveback_urb+0x60/0x8e [usbcore] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > qh_completions+0x91/0x3e9 [ehci_hcd] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > ehci_work+0x93/0x780 [ehci_hcd] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > ktime_get_ts+0x1d/0x3f >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> ktime_get+0xd/0x2d >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > getnstimeofday+0x4d/0xca >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > ehci_irq+0x147/0x197 [ehci_hcd] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> ktime_get+0xd/0x2d >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > usb_hcd_irq+0x24/0x58 [usbcore] >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > handle_IRQ_event+0x49/0xf8 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > handle_level_irq+0x50/0x85 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> handle_irq+0x17/0x1c >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> do_IRQ+0x2b/0x63 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > common_interrupt+0x29/0x30 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > audit_log_exit+0xb78/0xc8b >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > mwait_idle+0x75/0xa0 >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> cpu_idle+0x23/0x3f >> > Jun 3 15:03:52 kernel: [361033.894565] [] ? >> > start_kernel+0x262/0x265 >> > Jun 3 15:03:52 kernel: [361033.894565] ---[ end trace 9422ad58c50dc1ad >> ]--- >> > Jun 3 15:03:54 kernel: [361036.107365]* usb 5-3: USB disconnect, >> address 2 >> > * >> > Jun 3 15:03:54 kernel: [361036.112475] ERR-xpp_usb: xusb-0 >> > (usb-:00:1d.7-3) [X1047686]: nonzero write bul
