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Hi, My Asterisk CVS is apparently not doing much (other than keeping SIP & IAX2 registrations alive and doing some ZAP calls (without echo-cancellation), but slowly the memory is filling up, so much so that 100m virtual memory is used up within 12 hours and I have to restart the asterisk application every 48 hours to make sure I have enough memory... How can I help resolve this problem?

Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in

...-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 ? > > Do you mea...

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

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain

Hi, I am running Asterisk CVS from 8/27/04, and since about 8/17/04 Asterisk crashes on reload. I did remove support for h323 (as it crashes my * at random, and I don't need it currently). Here is a cut-out of the last lines when I give a reload command... == Parsing '/etc/asterisk/voicemail.conf': Found -- Reloading module 'app_queue.so' (True Call Queueing) ==

Hi, I am trying to install the h323 channel on the latest Asterisk CVS. I have re-compiled openh323 and pwlib, and now make in /usr/src/asterisk/channels/h323 works g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I../../include -I/usr/src/pwlib/include

>> >> After looking at the source I had also tried to increase the buffer size >> from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, >> and >> I still had the problem... >> >> I like speex and would like to use it (as I find ilbc a bit too scratchy) >> >> I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4

Hi, I am playing with conferencing, but might have hit a bug... Any use who wants to hang up or leave the conference should press the # key, after which they get a "goodbye" message and the call gets disconnected. However, this does not happen. whatever keys are pressed by whichever party gets heard on every other party. I am using Zap channels (Digium T405p) My extensions.conf

Hi, (sorry for my mistake in not deleting the rest of the message just now) The problem seems to be here in zaptel.c (and torisa.c) #ifdef DEFINE_SPINLOCK static DEFINE_SPINLOCK(zaptimerlock); static DEFINE_SPINLOCK(bigzaplock); #else static spinlock_t zaptimerlock = SPIN_LOCK_UNLOCKED; static spinlock_t bigzaplock = SPIN_LOCK_UNLOCKED; #endif If I remark out as follows: //#ifdef

Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting

Thanks Peter, Yes, indeed the 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

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

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

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion >

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, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

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