Hi Manfred, I applied the patch and recompiled and reinstalled and I got the folowing warning during my first test call: Aug 19 12:26:51 WARNING[294927]: 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-stuff.0.1.0-RC4) on a Debian Sarge box... > | I've got a generic HFC-S BRI ISDN card using i4l for inbound & outbound > calls > | (I gave up on getting zaphfc working). > | > | Asterisk is crashing out with a floating point exception a couple of > minutes > | into a SIP > PSTN call. SIP > SIP calls seem fine. Any idea where to > start > | with this issue? > | > | Gary > | > > Hi, > > I had the same problem. For me, the patch below works. It may not be the > correct way to solve this, though. > > Let me know, if it works also for you. > > Manfred > > > > > --- ../asterisk-v-1_0_RC2/dsp.c 2004-07-24 23:06:54.000000000 +0200 > +++ dsp.c 2004-08-18 10:11:26.017124954 +0200 > @@ -1229,6 +1229,13 @@ > int x; > int res = 0; > > + /*PM BEGIN*/ > + if (len==0) { > + ast_log(LOG_WARNING, "zero length packet\n"); > + return 0; > + } > + /*PM END*/ > + > accum = 0; > for (x=0;x<len; x++) > accum += abs(s[x]); > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
We're almost there.... the next problem is with the inbound calls over i4l. When the call is received, the "thank you for calling. press 1 for...." announcement is supposed to start... but something strange happens. The last fraction of a second of audio from the previous call plays first (weird or what?) and I get a warning on the console: Aug 19 15:06:20 NOTICE[491534]: channel.c:1650 ast_set_write_format: Unable to find a path from UNKN to SLINR I tried to use "monitor" to record the call but it just recorded the expected announcement, without the strange "echo". any ideas? Gary ----- Original Message ----- From: "Gary Pigott" <asterisk@garypigott.net> To: <asterisk-users@lists.digium.com> Sent: Thursday, August 19, 2004 12:59 PM Subject: Re: [Asterisk-Users] Floating point exception help> Hi Manfred, > > I applied the patch and recompiled and reinstalled and I got the folowing > warning during my first test call: > Aug 19 12:26:51 WARNING[294927]: 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-stuff.0.1.0-RC4) on a Debian Sarge box... >> | I've got a generic HFC-S BRI ISDN card using i4l for inbound & outbound >> calls >> | (I gave up on getting zaphfc working). >> | >> | Asterisk is crashing out with a floating point exception a couple of >> minutes >> | into a SIP > PSTN call. SIP > SIP calls seem fine. Any idea where to >> start >> | with this issue? >> | >> | Gary >> | >> >> Hi, >> >> I had the same problem. For me, the patch below works. It may not be the >> correct way to solve this, though. >> >> Let me know, if it works also for you. >> >> Manfred >> >> >> >> >> --- ../asterisk-v-1_0_RC2/dsp.c 2004-07-24 23:06:54.000000000 +0200 >> +++ dsp.c 2004-08-18 10:11:26.017124954 +0200 >> @@ -1229,6 +1229,13 @@ >> int x; >> int res = 0; >> >> + /*PM BEGIN*/ >> + if (len==0) { >> + ast_log(LOG_WARNING, "zero length packet\n"); >> + return 0; >> + } >> + /*PM END*/ >> + >> accum = 0; >> for (x=0;x<len; x++) >> accum += abs(s[x]); >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
I managed to record it by dialing out to the ISDN phone number. The recording (113KB) is at http://www.garypigott.net/files/asterisk/recording-in.wav Regards, Gary ----- Original Message ----- From: "Gary Pigott" <asterisk@garypigott.net> To: <asterisk-users@lists.digium.com> Sent: Thursday, August 19, 2004 3:12 PM Subject: Re: [Asterisk-Users] Floating point exception help> We're almost there.... the next problem is with the inbound calls over > i4l. > > When the call is received, the "thank you for calling. press 1 for...." > announcement is supposed to start... but something strange happens. The > last fraction of a second of audio from the previous call plays first > (weird or what?) and I get a warning on the console: > Aug 19 15:06:20 NOTICE[491534]: channel.c:1650 ast_set_write_format: > Unable to find a path from UNKN to SLINR > I tried to use "monitor" to record the call but it just recorded the > expected announcement, without the strange "echo". > > any ideas? > > Gary > > ----- Original Message ----- > From: "Gary Pigott" <asterisk@garypigott.net> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, August 19, 2004 12:59 PM > Subject: Re: [Asterisk-Users] Floating point exception help > > >> Hi Manfred, >> >> I applied the patch and recompiled and reinstalled and I got the folowing >> warning during my first test call: >> Aug 19 12:26:51 WARNING[294927]: 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-stuff.0.1.0-RC4) on a Debian Sarge box... >>> | I've got a generic HFC-S BRI ISDN card using i4l for inbound & >>> outbound calls >>> | (I gave up on getting zaphfc working). >>> | >>> | Asterisk is crashing out with a floating point exception a couple of >>> minutes >>> | into a SIP > PSTN call. SIP > SIP calls seem fine. Any idea where to >>> start >>> | with this issue? >>> | >>> | Gary >>> | >>> >>> Hi, >>> >>> I had the same problem. For me, the patch below works. It may not be the >>> correct way to solve this, though. >>> >>> Let me know, if it works also for you. >>> >>> Manfred >>> >>> >>> >>> >>> --- ../asterisk-v-1_0_RC2/dsp.c 2004-07-24 23:06:54.000000000 +0200 >>> +++ dsp.c 2004-08-18 10:11:26.017124954 +0200 >>> @@ -1229,6 +1229,13 @@ >>> int x; >>> int res = 0; >>> >>> + /*PM BEGIN*/ >>> + if (len==0) { >>> + ast_log(LOG_WARNING, "zero length packet\n"); >>> + return 0; >>> + } >>> + /*PM END*/ >>> + >>> accum = 0; >>> for (x=0;x<len; x++) >>> accum += abs(s[x]); >>> >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >