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...27]: dsp.c:1234 __ast_dsp_silence: zero length packet It looks like that could be the problem... and the fix! I'll let you know if the problem reoccurs. Might it be an idea to submit the patch to the bugtracker? Thanks, Gary ----- Original Message ----- From: "Manfred Petz" <pm@deuromedia.at> To: <asterisk-users@lists.digium.com> Sent: Thursday, August 19, 2004 12:08 PM Subject: Re: [Asterisk-Users] Floating point exception help > On Thu, 19 Aug 2004, Gary Pigott wrote: > > | I'm running a fresh install of * (CVS-HEAD-08/13/04 with bristuff from > | bri-st...

Hi, Is there a possibility to control sound levels (higher / lower) in Asterisk (so the codecs). Somebody asked me to evaluate that but I didn`t found any documentation about. I have the opinion that for these (audio) things the end user client is the only part where I can tune around. Problem is for example a (Austria) ISDN --> Asterisk --> SIP / IP ---> (Romania) Asterisk

Hi, Problem is: I have a Dell 1950 server with 6 NIC's ( 1 for Voice / Asterisk rest of them for other functions). The Voice LAN is on the 172.16.3.0 (255.255.0.0) subnet. One the other NICS there are different but also Class B like 172.15.1.x and so on. No problem at all i think. Now asterisk is up and running, I connected a Thomson 2030 directly to the NIC with an cross cable (to avoid

...twork topologies. My Asterisk version is the 1.4.12. Thank you and bye. Marco Signorini > ---------------------------- Original Message ---------------------------- > Subject: [asterisk-users] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11 > From: "Erik Wartusch" <we at deuromedia.at> > Date: Tue, November 13, 2007 10:25 am > To: asterisk-users at lists.digium.com > -------------------------------------------------------------------------- > > > Thx John !! > > Hmm I found now on voip-info.org a lot of Beta releases which should fix my &...

Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hi, Can anyone recommend a BRI card which works fine with asterisk and which supports point-to-point mode? Software fax detection should also work. Price does not matter. :) Digium seems to sell only PRI cards, and the Beronet drivers for the quad BRI cards seem to be in an early stage of development (besides, fax detection seems not to be implemented). Thanks pm

hi, i had asterisk and gnugk running on fedora core 2. it worked quite well. then, i needed to change to red hat 9, and i'm experiencing troubles with h.323 :-( making a call from a h.323 phone (innovaphone) does not work, and dial-in also doesn't. below is an excerpt of what happens, when i try to dial-in my extension (126). it takes about 10(!) seconds, until the 'Called 126'

Hi, I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed version ) to this version and in my opinion a lot more troubles arose.... For outgoing calls I use a Digium B410P with chan_misdn (before a Junghanns QuadBRI with zap). 1) So first thing is, that a user reports to me (highly

Hi, Im sure somebody out there had the same "problem before. IF a call comes in with suppressed caller id (Call Centers, etc.) 'asterisk' is shown as CallerID. Can I change somewhere this behaviour to display like ' Unknown' ? Thanks! Kind Regards, Erik

Hi, I've a huge problem with the following: Setup: Asterisk 1.4.11 I've got two Thomson ST2030s in an queue. After a while Asterisk logs the following if somebody calls the queues number: - Got SIP response 486 "Busy Here" back from 172.10.3.31 -- SIP/office1-0823d190 is busy -- Nobody picked up in 0 ms The phones are NOT busy (show channels show nothing). Also

Hi, I`m using several GXP2020 phones with newest Firmware 1.1.4.18. Asterisk Version: 1.4.11. It happens several times that users complain that the caller cannot hear the transmitted voice from the phones.... Also now it happens quite often that callers on hold beeing dropped. Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name (only IPS configured). I configured

Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest "official" release you can expect a few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped calls). So you have to install the BETAS whether you want or not...

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik