Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
arkda wrote: Nothing in the console aside from what I've posted. When a DTMF tone is played the server freezes instantly, hard reboot required. Just to close out this thread, it appears that this issue was related to http://bugs.digium.com/view.php?id=12053 Adding a loadzone and defaultzone to the /etc/zaptel.conf file resolved the server freeze/ crash. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
arkda wrote: Nothing in the console aside from what I've posted. When a DTMF tone is played the server freezes instantly, hard reboot required. Distribution is SuSE 10.2, kernel 2.6.18.8-0.7-default The actual dialplan on this server is very simple, only one phone and a few Dial commands via SIP to another server. I've disabled as much as possible (including an extra NIC which botched by G729 codecs) to try to eliminate any IRQ issues with no luck. I've tried using various codecs including ulaw with dtmfmode=inband, but each time the server freezes. Would you be willing to try another branch of zaptel that is currently in beta? I would be interested if this version results in a server lock up with your configuration. The beta version is at: http://svn.digium.com/svn/zaptel/team/sruffell/voicebus If you want / need any help with this beta version, please contact me directly. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
Sure Shaun, I'll give it a shot. I'll contact you directly to let you know the results. On Wed, Feb 27, 2008 at 10:33 AM, Shaun Ruffell [EMAIL PROTECTED] wrote: arkda wrote: Nothing in the console aside from what I've posted. When a DTMF tone is played the server freezes instantly, hard reboot required. Distribution is SuSE 10.2, kernel 2.6.18.8-0.7-default The actual dialplan on this server is very simple, only one phone and a few Dial commands via SIP to another server. I've disabled as much as possible (including an extra NIC which botched by G729 codecs) to try to eliminate any IRQ issues with no luck. I've tried using various codecs including ulaw with dtmfmode=inband, but each time the server freezes. Would you be willing to try another branch of zaptel that is currently in beta? I would be interested if this version results in a server lock up with your configuration. The beta version is at: http://svn.digium.com/svn/zaptel/team/sruffell/voicebus If you want / need any help with this beta version, please contact me directly. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below). When the server freezes there is no indication of any kind in the log on what the cause could be. When a call is made the following is logged on the console: [Feb 25 14:47:36] WARNING[3409]: chan_zap.c:1437 zt_enable_ec: Unable to enable echo cancellation on channel 1 (Argument list too long) -- Executing [EMAIL PROTECTED]:1] Goto(Zap/1-1, internal||1) in new stack -- Goto (internal,,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, SIP/user1) in new stack -- Called user1 -- SIP/user1-0820fcf8 is ringing ioctl(ZT_FREEZONE) failed: Inappropriate ioctl for device Failed to register zone 'France': No data available Not sure why it's listing France as the zone (this is in the US). I'm pretty new to Digium hardware so any advice is welcome. Below are the results of some of my testing. Am I crazy or is IRQ 217 not right? # cat /proc/interrupts CPU0 CPU1 0: 295139 0IO-APIC-edge timer 1: 8 0IO-APIC-edge i8042 6: 5 0IO-APIC-edge floppy 8: 2 0IO-APIC-edge rtc 9: 1 0 IO-APIC-level acpi 12:103 0IO-APIC-edge i8042 14: 12689 0IO-APIC-edge libata 15: 8792 0IO-APIC-edge libata 169: 3480 0 IO-APIC-level eth0 177: 0 0 IO-APIC-level ehci_hcd:usb1 185: 0 0 IO-APIC-level uhci_hcd:usb2 201: 0 0 IO-APIC-level uhci_hcd:usb3 217:1159061 0 IO-APIC-level wcte12x[p] NMI: 0 0 LOC: 294996 294975 ERR: 0 MIS: 0 # lspci -v 00:00.0 Host bridge: Intel Corporation 82875P/E7210 Memory Controller Hub (rev 02) Subsystem: IBM Unknown device 02ad Flags: bus master, fast devsel, latency 0 Memory at d200 (32-bit, prefetchable) [size=32M] Capabilities: [e4] Vendor Specific Information 00:03.0 PCI bridge: Intel Corporation 82875P/E7210 Processor to PCI to CSA Bridge (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, 66MHz, fast devsel, latency 48 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 2000-2fff Memory behind bridge: d010-d01f Prefetchable memory behind bridge: a800-a80f 00:1c.0 PCI bridge: Intel Corporation 6300ESB 64-bit PCI-X Bridge (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, 66MHz, fast devsel, latency 48 Bus: primary=00, secondary=03, subordinate=03, sec-latency=48 I/O behind bridge: 3000-3fff Memory behind bridge: d020-d02f Prefetchable memory behind bridge: a810-a81f Capabilities: [50] PCI-X bridge device 00:1d.0 USB Controller: Intel Corporation 6300ESB USB Universal Host Controller (rev 02) (prog-if 00 [UHCI]) Subsystem: IBM Unknown device 02ae Flags: bus master, medium devsel, latency 0, IRQ 185 I/O ports at 1400 [size=32] 00:1d.1 USB Controller: Intel Corporation 6300ESB USB Universal Host Controller (rev 02) (prog-if 00 [UHCI]) Subsystem: IBM Unknown device 02ae Flags: bus master, medium devsel, latency 0, IRQ 201 I/O ports at 1420 [size=32] 00:1d.4 System peripheral: Intel Corporation 6300ESB Watchdog Timer (rev 02) Subsystem: IBM Unknown device 02ae Flags: medium devsel Memory at d000 (32-bit, non-prefetchable) [size=16] 00:1d.5 PIC: Intel Corporation 6300ESB I/O Advanced Programmable Interrupt Controller (rev 02) (prog-if 20 [IO(X)-APIC]) Subsystem: IBM Unknown device 02ae Flags: bus master, fast devsel, latency 0 Capabilities: [50] PCI-X non-bridge device 00:1d.7 USB Controller: Intel Corporation 6300ESB USB2 Enhanced Host Controller (rev 02) (prog-if 20 [EHCI]) Subsystem: IBM Unknown device 02ae Flags: bus master, medium devsel, latency 0, IRQ 177 Memory at d400 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 0a) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=32 Memory behind bridge: d030-d03f Prefetchable memory behind bridge: e000-efff 00:1f.0 ISA bridge: Intel
Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote: Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below). When the server freezes there is no indication of any kind in the log on what the cause could be. When a call is made the following is logged on the console: Do you see anything in the console? What Linux distribution is it? What kerenl? To eliminate the option of a simple dialplan loop: try running Asterisk without -p . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
My guess is a mismatch between Asterisk, Zaptel, and libPRI. Make sure you are running the latest of each. Tzafrir Cohen wrote: On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote: Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below). When the server freezes there is no indication of any kind in the log on what the cause could be. When a call is made the following is logged on the console: Do you see anything in the console? What Linux distribution is it? What kerenl? To eliminate the option of a simple dialplan loop: try running Asterisk without -p . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
It's the latest from each (branch, not trunk): Asterisk revision 104093, zaptel revision 3849, libpri revision 529 all from svn. On Mon, Feb 25, 2008 at 3:50 PM, Eric Wieling [EMAIL PROTECTED] wrote: My guess is a mismatch between Asterisk, Zaptel, and libPRI. Make sure you are running the latest of each. Tzafrir Cohen wrote: On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote: Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below). When the server freezes there is no indication of any kind in the log on what the cause could be. When a call is made the following is logged on the console: Do you see anything in the console? What Linux distribution is it? What kerenl? To eliminate the option of a simple dialplan loop: try running Asterisk without -p . -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
Nothing in the console aside from what I've posted. When a DTMF tone is played the server freezes instantly, hard reboot required. Distribution is SuSE 10.2, kernel 2.6.18.8-0.7-default The actual dialplan on this server is very simple, only one phone and a few Dial commands via SIP to another server. I've disabled as much as possible (including an extra NIC which botched by G729 codecs) to try to eliminate any IRQ issues with no luck. I've tried using various codecs including ulaw with dtmfmode=inband, but each time the server freezes. On Mon, Feb 25, 2008 at 4:43 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote: Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below). When the server freezes there is no indication of any kind in the log on what the cause could be. When a call is made the following is logged on the console: Do you see anything in the console? What Linux distribution is it? What kerenl? To eliminate the option of a simple dialplan loop: try running Asterisk without -p . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
