Do you have a zap device for timing? On Wed, 2003-09-03 at 17:48, Zak wrote:> Hi, > > Every time I dial into my asterisk box i hear nothing but asterisk > jittering. > The following is an example of what I get on the asterisk CLI > > Thanks > > *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT > on RTP > to 0 > DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res > DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user > 'xirak' is 1 > out of 0 > DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: > Contact hop > : <sip:192.168.7.3> > -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack > DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format > changed from U > NKN to ULAW > DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling > timer at 16 > 0 sample intervals > -- Playing 'vm-login' > DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping > retransmission on > '6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found > DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling > timer at 0 > sample intervals > DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling > timer at 0 > sample intervals > WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): > Couldn't read u > sername > == Spawn extension (extensions, 1001, 1) exited non-zero on > 'SIP/xirak-259d' > DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak) > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Steven Critchfield <critch@basesys.com>
Hi,
Every time I dial into my asterisk box i hear nothing but asterisk
jittering.
The following is an example of what I get on the asterisk CLI
Thanks
*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT
on RTP
to 0
DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user
'xirak' is 1
out of 0
DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route:
Contact hop
: <sip:192.168.7.3>
-- Executing VoiceMailMain2("SIP/xirak-259d", "") in new
stack
DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format
changed from U
NKN to ULAW
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling
timer at 16
0 sample intervals
-- Playing 'vm-login'
DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping
retransmission on
'6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling
timer at 0
sample intervals
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling
timer at 0
sample intervals
WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain):
Couldn't read u
sername
== Spawn extension (extensions, 1001, 1) exited non-zero on
'SIP/xirak-259d'
DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak)
Hi Steven, I have a zap device installed in the box but I'm not sure if that's the one used for timing. thanks. Zak Subject: Re: [Asterisk-Users] Asterisk Jitters From: Steven Critchfield <critch@basesys.com> To: asterisk-users@lists.digium.com Date: Wed, 03 Sep 2003 11:17:15 -0500 Reply-To: asterisk-users@lists.digium.com Do you have a zap device for timing? On Wed, 2003-09-03 at 17:48, Zak wrote:>> Hi, >> >> Every time I dial into my asterisk box i hear nothing but asterisk >> jittering. >> The following is an example of what I get on the asterisk CLI >> >> Thanks >> >> *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT >> on RTP >> to 0 >> DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res >> DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user >> 'xirak' is 1 >> out of 0 >> DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: >> Contact hop >> : <sip:192.168.7.3> >> -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack >> DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format >> changed from U >> NKN to ULAW >> DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >> timer at 16 >> 0 sample intervals >> -- Playing 'vm-login' >> DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping >> retransmission on >> '6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found >> DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >> timer at 0 >> sample intervals >> DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >> timer at 0 >> sample intervals >> WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): >> Couldn't read u >> sername >> == Spawn extension (extensions, 1001, 1) exited non-zero on >> 'SIP/xirak-259d' >> DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak) >
I have three fxos from Digium installed in the box. The Box got Pentium 4 2.4 Ghz and 512 RAM. I had the box working fine once but it stopped working (jitters) after a reboot. #### If you have one, and the card is up and running, then it would be used for timing. Basically it is just needed in this case to make sure asterisk keeps chugging along at a known speed. What speed hardware are you using?>Date: Wed, 03 Sep 2003 21:05:04 -0700 >From: Zak <zakforever@netscape.net> >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Re: Asterisk Jitters >Reply-To: asterisk-users@lists.digium.com > > > >Hi Steven, > >I have a zap device installed in the box but I'm not sure if that's the one used for timing. > >thanks. > >Zak > > > >Subject: Re: [Asterisk-Users] Asterisk Jitters >From: Steven Critchfield <critch@basesys.com> >To: asterisk-users@lists.digium.com >Date: Wed, 03 Sep 2003 11:17:15 -0500 >Reply-To: asterisk-users@lists.digium.com > >Do you have a zap device for timing? > >On Wed, 2003-09-03 at 17:48, Zak wrote: > > > >>>Hi, >>> >>>Every time I dial into my asterisk box i hear nothing but asterisk >>>jittering. >>>The following is an example of what I get on the asterisk CLI >>> >>>Thanks >>> >>>*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT >>>on RTP >>>to 0 >>>DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res >>>DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user >>>'xirak' is 1 >>> out of 0 >>>DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: >>>Contact hop >>>: <sip:192.168.7.3> >>> -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack >>>DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format >>>changed from U >>>NKN to ULAW >>>DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >>>timer at 16 >>>0 sample intervals >>> -- Playing 'vm-login' >>>DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping >>>retransmission on >>>'6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found >>>DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >>>timer at 0 >>>sample intervals >>>DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >>>timer at 0 >>>sample intervals >>>WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): >>>Couldn't read u >>>sername >>> == Spawn extension (extensions, 1001, 1) exited non-zero on >>>'SIP/xirak-259d' >>>DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak) >>> >>> > > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030903/7ebe345e/attachment.htm
> >Message: 3 >Subject: Re: [Asterisk-Users] Re: Asterisk Jitters >From: Steven Critchfield <critch@basesys.com> >To: asterisk-users@lists.digium.com >Date: Wed, 03 Sep 2003 23:13:58 -0500 >Reply-To: asterisk-users@lists.digium.com > >On Thu, 2003-09-04 at 01:43, Zak wrote: > > >>I have three fxos from Digium installed in the box. >>The Box got Pentium 4 2.4 Ghz and 512 RAM. >>I had the box working fine once but it stopped working (jitters) after >>a reboot. >> >> > >Did you make sure the zap card drivers are loaded? Have you checked your >IRQs to make sure they didn't wander and start causing problems? >Steven, The zap card drivers are loaded correctly and there don't seem to be any IRQ problem. Here is what I get from /proc/interrupts. I'm not sure the sound card and eth0 using the same IRQ could be causing the problem? CPU0 0: 17728993 XT-PIC timer 1: 44037 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 9776462 XT-PIC eth0, Intel ICH2 8: 1 XT-PIC rtc 9: 103179821 XT-PIC wcfxo 10: 217202079 XT-PIC nvidia, wcfxo, wcfxo 12: 524504 XT-PIC PS/2 Mouse 14: 165140 XT-PIC ide0 15: 281208 XT-PIC ide1 NMI: 0 ERR: 0 Zak> > >>#### >>If you have one, and the card is up and running, then it would be used >>for timing. Basically it is just needed in this case to make sure >>asterisk keeps chugging along at a known speed. >> >>What speed hardware are you using? >> >> >>>Date: Wed, 03 Sep 2003 21:05:04 -0700 >>>From: Zak <zakforever@netscape.net> >>>To: asterisk-users@lists.digium.com >>>Subject: [Asterisk-Users] Re: Asterisk Jitters >>>Reply-To: asterisk-users@lists.digium.com >>> >>> >>>Hi Steven, >>> >>>I have a zap device installed in the box but I'm not sure if that's the one used for timing. >>> >>>thanks. >>> >>>Zak >>> >>> >>> >>>Subject: Re: [Asterisk-Users] Asterisk Jitters >>>From: Steven Critchfield <critch@basesys.com> >>>To: asterisk-users@lists.digium.com >>>Date: Wed, 03 Sep 2003 11:17:15 -0500 >>>Reply-To: asterisk-users@lists.digium.com >>> >>>Do you have a zap device for timing? >>> >>>On Wed, 2003-09-03 at 17:48, Zak wrote: >>> >>> >>> >>> >>>>>Hi, >>>>> >>>>>Every time I dial into my asterisk box i hear nothing but asterisk >>>>>jittering. >>>>>The following is an example of what I get on the asterisk CLI >>>>> >>>>>Thanks >>>>> >>>>>*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT >>>>>on RTP >>>>>to 0 >>>>>DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res >>>>>DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user >>>>>'xirak' is 1 >>>>> out of 0 >>>>>DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: >>>>>Contact hop >>>>>: <sip:192.168.7.3> >>>>> -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack >>>>>DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format >>>>>changed from U >>>>>NKN to ULAW >>>>>DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >>>>>timer at 16 >>>>>0 sample intervals >>>>> -- Playing 'vm-login' >>>>>DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping >>>>>retransmission on >>>>>'6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found >>>>>DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >>>>>timer at 0 >>>>>sample intervals >>>>>DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling >>>>>timer at 0 >>>>>sample intervals >>>>>WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): >>>>>Couldn't read u >>>>>sername >>>>> == Spawn extension (extensions, 1001, 1) exited non-zero on >>>>>'SIP/xirak-259d' >>>>>DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak) >>>>> >>>>> >>>>> >>> >>> >>> >>>-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030904/237bf54b/attachment.htm
Hi Steven. I have done as you suggested and I'm still getting the same problem. /proc/interrupts lists the following: 0: 45489 XT-PIC timer 1: 235 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 335816 XT-PIC wcfxo, Intel ICH2 8: 1 XT-PIC rtc 9: 0 XT-PIC usb-uhci 10: 829 XT-PIC eth0 11: 0 XT-PIC usb-uhci 12: 194 XT-PIC PS/2 Mouse 14: 4402 XT-PIC ide0 15: 2 XT-PIC ide1 NMI: 0 ERR: 0 I am also getting the following message when asterisk starts.. but I'm not sure if it means anything? WARNING[16384]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[114696]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailabl thank, Zak>Bing,Bing,Bing, we have the problem. nvidia and wcfxo cards on the same >interupt.>I'd say try removing a 2 WCFXO cards from the system and see if the >interupts free up, and your jitter stops.> 12: 524504 XT-PIC PS/2 Mouse > 14: 165140 XT-PIC ide0 > 15: 281208 XT-PIC ide1 > NMI: 0 > ERR: 0-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030905/ab1e2524/attachment.htm
>Message: 2 >Subject: Re: [Asterisk-Users] Re: Asterisk Jitters >From: Eric Wieling <eric@fnords.org> >To: asterisk-users@lists.digium.com >Date: Fri, 05 Sep 2003 11:54:19 -0500 >Reply-To: asterisk-users@lists.digium.com > >As you can see wcfxo is still sharing an IRQ. It won't work well if it >shares an IRQ. >I have changed the pci slot of the fxo so that it won't share IRQ with another device but the jittering is still there. check the interrupts list below CPU0 0: 1158022 XT-PIC timer 1: 807 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 1377109 XT-PIC eth0, Intel ICH2 8: 1 XT-PIC rtc 10: 12231674 XT-PIC wcfxo 12: 14239 XT-PIC PS/2 Mouse 14: 18691 XT-PIC ide0 15: 21804 XT-PIC ide1 NMI: 0 ERR: 0>On Fri, 2003-09-05 at 19:39, Zak wrote: > > >>Hi Steven. >> >>I have done as you suggested and I'm still getting the same problem. >>/proc/interrupts lists the following: >> >>0: 45489 XT-PIC timer >> 1: 235 XT-PIC keyboard >> 2: 0 XT-PIC cascade >> 5: 335816 XT-PIC wcfxo, Intel ICH2 >> 8: 1 XT-PIC rtc >> 9: 0 XT-PIC usb-uhci >> 10: 829 XT-PIC eth0 >> 11: 0 XT-PIC usb-uhci >> 12: 194 XT-PIC PS/2 Mouse >> 14: 4402 XT-PIC ide0 >> 15: 2 XT-PIC ide1 >>NMI: 0 >>ERR: 0 >> >>I am also getting the following message when asterisk starts.. but I'm >>not sure if it means anything? >> >>WARNING[16384]: File chan_oss.c, Line 974 (load_module): XXX I don't >>work right >>with non-full >>duplex sound cards XXX >> == Registered channel type 'Console' (OSS Console Channel Driver) >> == Parsing '/etc/asterisk/oss.conf': Found >>WARNING[114696]: File chan_oss.c, Line 232 (sound_thread): Read error >>on sound >>device: Resource temporarily unavailabl >> >>thank, >> >>Zak >> >> >>>Bing,Bing,Bing, we have the problem. nvidia and wcfxo cards on the same >>>interupt. >>> >>> >>>I'd say try removing a 2 WCFXO cards from the system and see if the >>>interupts free up, and your jitter stops. >>> >>> >>> 12: 524504 XT-PIC PS/2 Mouse >>> 14: 165140 XT-PIC ide0 >>> 15: 281208 XT-PIC ide1 >>>NMI: 0 >>>ERR: 0 >>> >>>
I have been reading archive post in regards to h323 support, and I am not clear on this: 1. Is h323 enabled and ready to use if one compiles asterisk, zaptel and libpri (as shown on asterisk.org) ? 2. If it is, is it h323 or oh323? 3. If it is not, does one just need to follow instructions in /asterisk/channels/h323/ in order to get it enabled. 4. I have an Ericsson webswitch 100 G4 (4 FXO ports). Once h323 is enabled/configured in *, WHY there is no any authentication needed on Ericsson box for placing the calls? I presume, that when Ericsson, "assigns" LAN IPs, that only those boxes can communicate with it? Is that correct? Thanks Senad
> On Fri, 5 Sep 2003, Zak wrote: > > > >>> I have changed the pci slot of the fxo so that it won't share IRQ with >>> another device but the jittering is still >>> there. check the interrupts list below >>> >>> CPU0 >>> 0: 1158022 XT-PIC timer >>> 1: 807 XT-PIC keyboard >>> 2: 0 XT-PIC cascade >>> 5: 1377109 XT-PIC eth0, Intel ICH2 >>> 8: 1 XT-PIC rtc >>> 10: 12231674 XT-PIC wcfxo >>> 12: 14239 XT-PIC PS/2 Mouse >>> 14: 18691 XT-PIC ide0 >>> 15: 21804 XT-PIC ide1 >>> NMI: 0 >>> ERR: 0 >> >> > >okay, now get APIC and get rid of the XT-PIC >and then we can start looking at why out jitter, is it on all zap ports? >is it continous? or only comes in occasionally? > > -wasim >I have not been able to get APIC working but from what I have seen , other machines with XT-PIC work perfectly. The thing to note is when I first installed asterisk, I had the same problem of jittering untill someone helped me fix the problem and everything used to work fine, and now the same problem is back. The jittering is continous and I can't make any kind of call. Does anyone have a clue? I can send my config files if that can help. Zak