Re: [asterisk-users] asterisk-users Digest, Vol 99, Issue 9
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom PSTN gateway - pstn line to telcoi'm using xlite for windows when I make a phone call (sip - outgoing channel),I can hear my own voice so clear. it's very annoying mewhen talking a little loud... any solution? Two questions: (1) Does the problem occur when you make a SIP-to-SIP call, without the PSTN being involved? No, it's happened only when I make a call from sip to pstn line. (2) When you hear your own voice in the headset, is it delayed, or is just an immediate louder-than-you-want side-tone? it's immediate voice and very clear, just like talk-to-my-ear with no delay If it *does* occur in SIP-to-SIP calls, this would rule out your XORCOM and the PSTN as the cause. If it's only occurring in SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction between them) is a likely suspect. There are several things which can cause this sort of problem. (A) Direct acoustic feedback within the headset. In this case, you'd probably hear it even if the headset was unplugged entirely. The only cure is to buy a better headset. (B) Incorrect audio-mixer settings in your PC. To the PC audio infrastructure, a headset usually looks like a microphone and a separate speaker. The audio mixer (hardware and software) usually has an ability to mix some of what the microphone hears into the speaker output. If this knob is turned up too high, you'll hear your own voice too loudly. If too low, you won't hear your own voice at all when you speak into the headset, and many people find this lack of side-tone to be confusing. The cure here is to adjust the audio side-tone level, either in your Windows audio-mixer control panel, or in X-Lite (if it has such an adjustment). (C) Electrical reflection from an analog impedance discontinuity in the analog telephone-line system. This can result from a mismatch between the telephone wiring, and the PSTN interface device, and can occur at any point in the analog transmission. If the loud side-tone you hear is *not* delayed noticeably, then the impedance mismatch might be at your XORCOM/PSTN interface. The XORCOM may have a software adjustment or jumper setting, to match its audio impedance to that of your local phone line... try fiddling with these settings to see if they reduce the excessive side-tone level. If the loud side-tone you hear is delayed (it sounds a bit like an echo) then it may very well be at the far end of the phone line, outside of your own physical control... it might be at your local phone office, or anywhere between you and the far end of the phone connection. Not much you can do about this. (D) Audio feedback at the far end of the call, in a cheap phone handset. Sometimes, audio from the back side of the speaker in a handset travels through the body of the handset and is picked up by the microphone, and results in an audible delayed echo of the voice from the far end of the line. Using a better handset, or stuffing the handset full of audio damping material (cloth or cotton or fiberglass) is the cure here. Well, thanks a lot Lee for suggestion and explanation, I'll try this tommorow. We've often faced this problem with SIP soft phones when the computer's sound system gain was set too high. You usually have to play around with microphone gain settings to get to the point where the echo disappears with the other party still being able to hear you. And thanks for your share Raj, I appreciate that.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I can hear my own voice through the headset
Sorry for my last post, Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom PSTN gateway - pstn line to telcoi'm using xlite for windows when I make a phone call (sip - outgoing channel),I can hear my own voice so clear. it's very annoying mewhen talking a little loud... any solution? Two questions: (1) Does the problem occur when you make a SIP-to-SIP call, without the PSTN being involved? No, it's happened only when I make a call from sip to pstn line. (2) When you hear your own voice in the headset, is it delayed, or is just an immediate louder-than-you-want side-tone? it's immediate voice and very clear, just like talk-to-my-ear with no delay If it *does* occur in SIP-to-SIP calls, this would rule out your XORCOM and the PSTN as the cause. If it's only occurring in SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction between them) is a likely suspect. There are several things which can cause this sort of problem. (A) Direct acoustic feedback within the headset. In this case, you'd probably hear it even if the headset was unplugged entirely. The only cure is to buy a better headset. (B) Incorrect audio-mixer settings in your PC. To the PC audio infrastructure, a headset usually looks like a microphone and a separate speaker. The audio mixer (hardware and software) usually has an ability to mix some of what the microphone hears into the speaker output. If this knob is turned up too high, you'll hear your own voice too loudly. If too low, you won't hear your own voice at all when you speak into the headset, and many people find this lack of side-tone to be confusing. The cure here is to adjust the audio side-tone level, either in your Windows audio-mixer control panel, or in X-Lite (if it has such an adjustment). (C) Electrical reflection from an analog impedance discontinuity in the analog telephone-line system. This can result from a mismatch between the telephone wiring, and the PSTN interface device, and can occur at any point in the analog transmission. If the loud side-tone you hear is *not* delayed noticeably, then the impedance mismatch might be at your XORCOM/PSTN interface. The XORCOM may have a software adjustment or jumper setting, to match its audio impedance to that of your local phone line... try fiddling with these settings to see if they reduce the excessive side-tone level. If the loud side-tone you hear is delayed (it sounds a bit like an echo) then it may very well be at the far end of the phone line, outside of your own physical control... it might be at your local phone office, or anywhere between you and the far end of the phone connection. Not much you can do about this. (D) Audio feedback at the far end of the call, in a cheap phone handset. Sometimes, audio from the back side of the speaker in a handset travels through the body of the handset and is picked up by the microphone, and results in an audible delayed echo of the voice from the far end of the line. Using a better handset, or stuffing the handset full of audio damping material (cloth or cotton or fiberglass) is the cure here. Well, thanks a lot Lee for suggestion and explanation, I'll try this tommorow. We've often faced this problem with SIP soft phones when the computer's sound system gain was set too high. You usually have to play around with microphone gain settings to get to the point where the echo disappears with the other party still being able to hear you. And thanks for your share Raj, I appreciate that.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I can hear my own voice through the headset
Hi all, Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom PSTN gateway - pstn line to telcoi'm using xlite for windows when I make a phone call (sip - outgoing channel),I can hear my own voice so clear. it's very annoying mewhen talking a little loud... any solution? Thanks, Frangky -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID problem with astribank
Hi all... I'm sorry for repeating my message. I have a problem with caller id on my asterisk server with xorcom astribank. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id. here is the log : -- Starting simple switch on 'DAHDI/15-1' [Apr 30 11:24:40] ERROR[30895]: callerid.c:562 callerid_feed: No start bit found in fsk data. [Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8712 ss_thread: CallerID feed failed: Success [Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8816 ss_thread: CallerID returned with error on channel 'DAHDI/15-1' Please help me _ NEW! Get Windows Live FREE. http://www.get.live.com/wl/all-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID on Asterisk and Astribank
Hi all... I have a problem with caller id on my asterisk server. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id. My extensions.conf : [incoming1] exten = s,1,AGI(/var/apps/core/runagi,incoming,${CALLERID(num)}) exten = s,n,QUEUE(${que},trkd) exten = h,1,Hangup() here is the log : -- Starting simple switch on 'DAHDI/15-1' [Apr 30 11:24:40] ERROR[30895]: callerid.c:562 callerid_feed: No start bit found in fsk data. [Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8712 ss_thread: CallerID feed failed: Success [Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8816 ss_thread: CallerID returned with error on channel 'DAHDI/15-1' == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's' == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to context 'default' -- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack -- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack -- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new stack -- Goto (im-incoming,s,1) -- Executing [...@im-incoming:1] AGI(DAHDI/15-1, /var/apps/core/runagi,incoming,) in new stack -- Launched AGI Script /var/apps/core/runagi -- Playing 'en/0006' (escape_digits=) (sample_offset 0) I read instructions from a few forums then I made a change on 'chan_dahdi.conf' like : - 1: cidsignalling=v23, cidstart=ring, hidecallerid=no, callerid=asreceived Here's the log : -- Starting simple switch on 'DAHDI/15-1' [Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18 (Ring Begin)... [Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 2 (Ring/Answered)... [Apr 30 11:42:05] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18 (Ring Begin)... == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's' == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to context 'default' -- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack -- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack [Apr 30 11:42:06] WARNING[31296]: chan_dahdi.c:6174 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 15 == Spawn extension (default, s, 2) exited non-zero on 'DAHDI/15-1' - 2: cidsignalling=dtmf', cidstart=ring, hidecallerid=no, callerid=asreceived Here's the log : -- Starting simple switch on 'DAHDI/15-1' [Apr 30 11:49:28] WARNING[31491]: chan_dahdi.c:8610 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch -- Hungup 'DAHDI/15-1' -- Starting simple switch on 'DAHDI/15-1' [Apr 30 11:49:34] DEBUG[31492]: chan_dahdi.c:8630 ss_thread: CID is '', flags 8 == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's' == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to context 'default' -- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack -- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack [Apr 30 11:49:35] WARNING[31492]: chan_dahdi.c:6174 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 15 -- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new stack -- Goto (im-incoming,s,1) -- Executing [...@im-incoming:1] AGI(DAHDI/15-1, /var/apps/core/runagi,incoming,) in new stack -- Launched AGI Script /var/apps/core/runagi - 3: cidsignalling=dtmf', cidstart=polarity, hidecallerid=no, callerid=asreceived Here's the log : -- Starting simple switch on 'DAHDI/15-1' == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's' == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to context 'default' -- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack -- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack -- Registered IAX2 '9009' (AUTHENTICATED) at 127.0.0.1:48961 -- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new stack -- Goto (im-incoming,s,1) -- Executing [...@im-incoming:1] AGI(DAHDI/15-1, /var/apps/core/runagi,incoming,) in new stack -- Launched AGI Script /var/apps/core/runagi - 4: cidsignalling=v23', cidstart=polarity, hidecallerid=no, callerid=asreceived Here's the log : -- Starting simple switch on 'DAHDI/15-1' == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's' == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to context 'default' -- Executing [...@default:1]
Re: [asterisk-users] Astribank problem
Ok, the problem solved... thanks for your advice. after rebooting i run /usr/share/dahdi/xpp_fxloader load and everything run normally. thanks... Message: 24 Date: Mon, 1 Feb 2010 11:03:44 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] Astribank problem To: asterisk-users@lists.digium.com Message-ID: 20100201090344.gr3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Mon, Feb 01, 2010 at 07:42:51AM +, frangky robert wrote: I do some test: 1.unplug usb connector from server to astricon 2.unplug power to astricon 3.plug-in the power to astricon 4.plug-in the usb connector Here is the log from /var/log/messages after doing the 1st step. Feb 1 19:38:24 localhost last message repeated 2 times Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: nonzero write bulk status received: -71 (pending_writes=1) Feb 1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3 Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: usb_submit_urb failed: -19 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Disconnecting Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Deactivating Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release XPDS Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove Feb 1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: worker(XBUS-00)-xpds_init_done=0 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Atribank Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Astribank Release Feb 1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: now disconnected Feb 1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 from /etc/dahdi/xpp_order Feb 1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline And, this is the log after doing 4th step. Feb 1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using ehci_hcd and address 4 Feb 1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 choice Feb 1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... XPP_HOTPLUG_DISABLED Seems like you explicitly disabled firmware loading by setting XPP_HOTPLUG_DISABLED in /etc/dahdi/init.conf . Just rem-out that line. lsusb result is: [r...@localhost ~]# lsusb Bus 002 Device 004: ID e4e4:1160 Bus 002 Device 001: ID : Bus 006 Device 001: ID : Bus 006 Device 002: ID 04b3:3025 IBM Corp. Bus 004 Device 001: ID : Bus 008 Device 001: ID : Bus 007 Device 001: ID : Bus 001 Device 001: ID : Bus 005 Device 001: ID : Bus 003 Device 001: ID : here is the msg when i do /usr/share/dahdi/xpp_fxloader [r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb This only runs USB firmware loading. And as the firmware loading is explicitly disabled on your system, the FPGA firmware will still not get loaded. This is also something that you would have seen if you would run 'dahdi_hardware -v' So basically just remove that line from init.conf and replug the Astribank. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir * _ New Windows 7: Find the right PC for you. Learn more. http://windows.microsoft.com/shop-- _ -- 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] Astribank problem
I do some test: 1.unplug usb connector from server to astricon 2.unplug power to astricon 3.plug-in the power to astricon 4.plug-in the usb connector Here is the log from /var/log/messages after doing the 1st step. Feb 1 19:38:24 localhost last message repeated 2 times Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: nonzero write bulk status received: -71 (pending_writes=1) Feb 1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3 Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: usb_submit_urb failed: -19 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Disconnecting Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Deactivating Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release XPDS Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove Feb 1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: worker(XBUS-00)-xpds_init_done=0 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Atribank Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Astribank Release Feb 1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: now disconnected Feb 1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 from /etc/dahdi/xpp_order Feb 1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline And, this is the log after doing 4th step. Feb 1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using ehci_hcd and address 4 Feb 1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 choice Feb 1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... XPP_HOTPLUG_DISABLED lsusb result is: [r...@localhost ~]# lsusb Bus 002 Device 004: ID e4e4:1160 Bus 002 Device 001: ID : Bus 006 Device 001: ID : Bus 006 Device 002: ID 04b3:3025 IBM Corp. Bus 004 Device 001: ID : Bus 008 Device 001: ID : Bus 007 Device 001: ID : Bus 001 Device 001: ID : Bus 005 Device 001: ID : Bus 003 Device 001: ID : here is the msg when i do /usr/share/dahdi/xpp_fxloader [r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[3955]: - FIRMWARE LOADING: (usb) [1 devices] Got all 1 devices 'xpp_fxloader'[4074]: - FIRMWARE IS LOADED but, when i do /etc/init.d/dahdi stop and start here is the result [r...@localhost ~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: xpp_usb: [ OK ] Astribanks initialization is starting Astribanks detection ..TIMEOUT No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] On Sat, Jan 30, 2010 at 03:57:30AM +, frangky robert wrote: H all... I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. My problem is, every time i unplug the astribank power supply, and reconnect it, astribank cannot work again (lsusb result is 11x0)... When you plug the Astribank, the firmware should get loaded by a script called from udev. What messages do you see in /var/log/messages following that? but, after reinstall the asterisk and dahdi, astribank will detected (lsusb result is 11x2)... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-id:SI_SB_3:092010-- _ -- 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] Astribank problem
H all... I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. My problem is, every time i unplug the astribank power supply, and reconnect it, astribank cannot work again (lsusb result is 11x0)... but, after reinstall the asterisk and dahdi, astribank will detected (lsusb result is 11x2)... any suggestion? Regard, frank. _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-id:SI_SB_3:092010-- _ -- 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] memory leak on asterisk 1.6.0.6
for the second time i'm asking in this forum, somebody help me my asterisk box have a problem with memory leak. I'm scheduling to rstart the box to fix this problem but any cleverer suggest to fix this? coz this issue causing another problem to my AGI application... thankyou before _ NEW! Get Windows Live FREE. http://www.get.live.com/wl/all___ -- 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] Memory leak on asterisk 1.6.0.6
Hi everyones, I have a production server using asterisk 1.6.0.6 using php as an IVR and mssql server (on other machine) My server attached a Sangoma A104 card (4xT1 card) i have a problem with memory leak on that server and causing a delay on IVR prompt. (Thats my assumption, memory leak problem) any suggestion on this problem? or maybe any assumption? frangky _ Join the Fantasy Football club and win cash prizes here! http://fantasyfootball.id.msn.com___ -- 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