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...It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-7...

...0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp Hope we can fix this somehow. Walter Klomp Singapore.

...en') == Spawn extension (default, 4084, 2) exited non-zero on 'SIP/67.23.212.25-0814f830' I have dropped this question at the asterisk user list some days ago, but it's being ignored... (or nobody has the answer) Can anybody shed some light on this ? Warmest Regards, Walter Klomp

...elaxdtmf=yes rxgain=0.0 txgain=2.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 context=default signalling=pri_net; this is connected to voice switch channel => 1-15 channel => 17-31 Any suggestions and assistance would be very welcome. Thanks in advance Walter Klomp Singapore. > >

...problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. On Thu, 29 Jul 2004, Walter Klomp wrote: > However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN > phone, the call does not get disconnected. My SIP phone goes quiet but > doesn't disconnect. If I a few seconds later pick up the PSTN phone again, > the connection is still there. Only if I...

Hi, After only 4 days I have 107472352 bytes in 46007 allocations in file 'pbx_spool.c' asterisk*CLI> show memory summary 180 bytes in 2 allocations in file 'netsock.c' 12 bytes in 1 allocations in file 'devicestate.c' 2268 bytes in 1 allocations in file 'jitterbuf.c' 8160 bytes in 1 allocations in file

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

...AWS Virtual Private Cloud or something similar and practice deploying freeipa a few times with a few clients. It takes some understanding of the caveats and implimentation before you will be able to deploy it successfully in a production environment. Good Luck! On 24 February 2015 at 01:40, Jitse Klomp <jitseklomp at gmail.com> wrote: > On 02/24/2015 01:15 AM, Gordon Messmer wrote: > >> On 02/23/2015 08:22 AM, Niki Kovacs wrote: >> >>> 1. Users should be manageable through a GUI, probably a web interface, >>> so the client can create, manage and delete t...

...t signalling=pri_cpe channel => 63-77 channel => 79-93 ; Span 4 group=4 context=default signalling=pri_net channel => 94-108 channel => 110-124 Hope anyone can shed some light on this. I have been breaking my head on this about 4 days now, trying just about anything... Thanks Walter Klomp Singapore.

...11:05:26, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ========================================================================= Connected to Asterisk CVS-HEAD-06/28/04-11:05:26 currently running on gk2 (pid = 15564) Is this normal ? Warmest Regards, Walter Klomp

...dr_in bindaddr' defined but not used ar cr libchanh323.a ast_h323.o But after I install the asterisk the chan_h323.so is not installed, and I see it's also not in the channels/h323 folder. Am I missing something ? I'm sure I read all the readme files. Hope somebody can help Walter Klomp Singapore -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041007/5a5a2d9c/attachment.htm

...l compile errors on x86_64 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1142417917.22407.0.camel@RobinHood.LinuxAutrement.local> > Content-Type: text/plain > > On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote: >> Hi, >> >> Just downloaded the latest cvs from zaptel on my sparking new Athlon64 >> Centos4.2 system, but hitting a stumbling block... (sorry for the long >> post) > > Kernel source installed? > -- > Dave Cotton <dcotton@linuxautrement.com&gt...

...4 4 6 4 3 - 17 - 25 G729A - 7 4 4 6 4 3 12 - - 25 SPEEX - - - - - - - - - - - ILBC - 8 5 5 7 5 4 13 18 - - Warmest Regards, Walter Klomp

Hi, I'm sorry to bring this up again, but I have been googling forever and whatever solutions are offered don't work for me. I am using x-lite (the latest build) and trying to use Speex. When I do call from the x-lite to another SIP phone or PSTN (through Cisco gateway) My asterisk fills up with this message: WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space The

...- - 25 SPEEX - - - - - - - - - - - ILBC - 8 5 5 7 5 4 13 18 - - !!! Help Anybody??? Digium ??? I need this G.729 to work as G.711 is too much a bandwidth hog and my Cisco 5300 is dying... Walter Klomp

On 02/24/2015 01:15 AM, Gordon Messmer wrote: > On 02/23/2015 08:22 AM, Niki Kovacs wrote: >> 1. Users should be manageable through a GUI, probably a web interface, >> so the client can create, manage and delete them eventually. > > FreeIPA is a good option, generally. As best I understand it, it's > currently available in a Docker container for CentOS. >

...charset=ISO-8859-1 Hi Walter, I had high load and extreme memory usage on my machine. My machine wasn't running on SMP. My point was that the cvs version you were using contained some bad patches, and it was probably a good idea to upgrade or move to stable. Thanks, Erik On 7/18/05, Walter Klomp <walter@aglow.com.sg> wrote: > Hi Erik, > > You put me to a page which refers to high load on CPU on SMP. Nothing to do > with memory leak. Furthermore I am not running SMP. > > Any other suggestions in which direction to look? Am I the only one > experiencing this ? &gt...

Most Linux distributions still compile against the original x86-64 baseline that was based on the AMD K8 (minus the 3DNow! parts, for Intel EM64T compatibility). There has been an attempt to use the existing AT_PLATFORM-based loading mechanism in the glibc dynamic linker to enable a selection of optimized libraries. But the general selection mechanism in glibc is problematic: hwcaps

Hi, I have a Cisco 5300 which I want to make a call THROUGH the Asterisk PBX (security) to an IP phone which supports g729, and vice versa. Both Cisco and the phone talk this codec if I do not force the call to go through * However if I say canreinvite=no in the sip.conf for either of these gadgets, the call will fail with No compatible codecs! I have bought a 5 user license just to

Hi, I was trying to get the fax capability for Asterisk, but the opencall.org nameservers don't seem to work anymore. Does anybody have the direct IP link to the source so I can get it from there ? Thanks Walter