Re: [Asterisk-Users] Cisco 7940 - Disappearing Clock
The clock on cisco phones 'disappears' when it fails to receive updates from the ntp server. This is most likely due to your ntp server configuration. By default the ntp mode on your cisco phone is directedbroadcast. If your ntp server doesn't support this you will need to change the mode on your phone to unicast. This is well documented by Cisco - Cisco SIP IP Phone Administrator Guide http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_book09186a00801d1978.html Please read! Cheers Sophus ___ 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] Echo Madness
Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine 1 x AMD A64/3500+ CPU: Desktop Athlon64 Retail w/fan SKT 1 x Asus A8N-SLI Deluxe Athlon 64 S939 NVIDIA nForce(r)4 SLI PCI Express Req: 24pin ATX 1 x Corsair TWINX1024-3200XL 2x 512MB 1024MB Dual Channel Optimised Pair 2 x Western Digital WD1200JD/PD-SATA 120GB Serial ATA 150Mbps 7200rpm HDD 8Mb Cache 1 x Leadtek PX6200 TC TDH GeForce 6200 64MB PCI-E 1 x LG GDR-8163B DVD-ROM Internal: 16x Black 1 x Alps FDD B Floppy Disk Drive: 1.44Mb 3.5 Black 1 x Antec 1040B ATX Full Tower Case: 400W PSU Black 1 x Generic Adapter: Power ATX 20 pin - 24 pin With a Digium Wildcard TDM400P TMD04B (4 Port FXO Modules) connecting to our analogue system. On calls that use the TMD400P we're getting echo on the IP phone - on inbound calls only. That is if a call is initiated from an IP phone there is no echo on either side of the conversation, but if the call originates from the PSTN the person on the IP phone gets echo for about the first 10-30 seconds of the call. IP to IP calls are echo free. Let me elaborate After installing Asterisk we had major echo problems when using the TMD400P. After tweaking the settings in the relevant configuration files the echo, the best result I could get was echo disappearing about 10-30 seconds into a call (on ALL calls, both incoming and outgoing). The settings I'm currently using are zaptel.conf defaultzone=au loadzone=au fxsks=1-4 zapata.conf [channels] language=en context=default rxwink=300 usecallerid=no hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=6.0 ; about 12 seems to effect echo but this level (6) is required for normal comms. txgain=-1.0 ; this seems the best level. switchtype=national group=1 callgroup=1 pickupgroup=1 immediate=no usecallerid=no flash=100 signalling=fxs_ks echocancel=128 echocancelwhenbridged=yes echotraining=900 callerid=asreceived busydetect=yes busycount=5 context=default channel = 1-4 I have tried practically every different configuration possible, and this seems to yield the best result for me, although not good enough for a production environment (I hate echo). Further reading on the mailing list and wiki suggested I should try tweaking a) In chan_zap.c - change the following line: #define READ_SIZE 160 to #define READ_SIZE 16 And In zapata.conf adding jitterbuffers=40 (and recompile) This did not have any noticeable effect, except increase CPU load. b) In zconfig.h, uncomment DAGGRESSIVE_SUPPRESSOR (and recompile) This made audio quality worse, and echo unchanged. My TDM400P card, which I purchased not two weeks ago is marked 'REF F' and the FXO module are marked 'REV B'. Discussion in the mailing lists suggest the Current card revision is H and the current FXO module revision is C. Does anyone know if the revision of card I'm using is more susceptible to echo problems? A dmesg also shows my card as a E/F REV. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI ACPI: PCI interrupt :05:08.0[A] - GSI 3 (level, low) - IRQ 3 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) A cat proc /interrupts shows CPU0 0: 13097582 XT-PIC timer 1:958 XT-PIC i8042 2: 0 XT-PIC cascade 3: 13742817 XT-PIC wctdm 4:3258008 XT-PIC NVidia CK804 5: 12081 XT-PIC libata, ehci_hcd 7: 3 XT-PIC SysKonnect SK-98xx 8: 0 XT-PIC rtc 9: 0 XT-PIC acpi 10: 0 XT-PIC libata, ohci_hcd 11:1323944 XT-PIC libata, eth0 12: 20212 XT-PIC i8042 15: 117438 XT-PIC ide1 NMI: 2146 LOC: 13095854 ERR: 0 MIS: 0 As digium recommends, wctdm has it's own interrupt. Also, I have tried moving my TDM400P Card to another pci slot there was no difference. I was stumped. Didn't know what to do. So, with echo still on my calls (both incoming and outgoing calls) for about 30 seconds I gave up and started playing with dialplans. The TDM400P card in my asterisk box is connected to an analogue extension port in my key system (I have also tried connecting it directly to the PSTN to see if the echo problems change with nothing positive to report), and as a result I need to dial a '0' followed by a pause to get an outside line. At this time I just had an entry in my
Re: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hi, I understand now that asterisk is waiting for callerid info, is it possible to stop asterisk waiting for this and just answer immediately? cheers Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hello, As per http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20zapata.conf I have set usecallerid=no. This has not made asterisk pickup immediately. I'd appreciate and advice... cheers Adam On Wed, 10 Nov 2004 08:31:19 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Thu, 2004-11-11 at 00:55 +1100, Sophus wrote: Hi, I understand now that asterisk is waiting for callerid info, is it possible to stop asterisk waiting for this and just answer immediately? Depends on if it is possible for you to read and edit the config file. Please look there and see how nicely it is documented. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hello, Ok... After trying all this stuff, what still confuses me is - I dial the zap interface -- Asterisk instantly says- --Starting simple switch on 'Zap/1-1' then a couple of seconds later Nov 4 19:11:39 NOTICE[1146895]: chan_zap.c:5356 ss_thread: Got event 2 (Ring/Answered)... then another couple of seconds later Nov 4 19:11:42 NOTICE[1146895]: chan_zap.c:5356 ss_thread: Got event 2 (Ring/Answered)... then another couple of seconds later -- Executing Answer(Zap/1-1, ) in new stack why all the delay in executing the answer? cheers Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hello, Ok... After trying all this stuff, what still confuses me is - I dial the zap interface -- Asterisk instantly says- --Starting simple switch on 'Zap/1-1' then a couple of seconds later Nov 4 19:11:39 NOTICE[1146895]: chan_zap.c:5356 ss_thread: Got event 2 (Ring/Answered)... then another couple of seconds later Nov 4 19:11:42 NOTICE[1146895]: chan_zap.c:5356 ss_thread: Got event 2 (Ring/Answered)... then another couple of seconds later -- Executing Answer(Zap/1-1, ) in new stack why all the delay in executing the answer? cheers Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hi, I set callerid=no in zapata.conf and exten = s,1,Wait(0) exten = s,2,Answer in extensions.conf but there is still a delay of about 5-8 seconds before asterisk picks up. any ideas? cheers adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hi, is it possible to change the amount of time it takes asterisk to pickup an incoming call on a zaptel interface? cheers Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAP Hook flash / recall on active zap interface
Hi there, thanks for that. I'm using x-lite softphone, can I send a flash to the x100p from xlite directly? or do I need to dial an extension and have asterisk do it? any examples...? cheers Adam On Thu, 16 Sep 2004 10:14:39 -0300, Marcelo Pacheco [EMAIL PROTECTED] wrote: The flash application flashes que current ZAP device, it doesn't take a parameter AFAIK. Correct me if I'm wrong. For instance, you could use this for manual call deflection on a POTS line, if the user asks to be transfered to another branch, you could use Flash then Dial the DTMF digits to call the other extension and transfer. Marcelo Pacheco Em Qui 16 Set 2004 07:25, Sophus escreveu: Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 presto we're on a three way chat, with me only using one line - using the telephone company's 3waychat feature... I have tried Flash(Zap/1), and similar commands, however an error is returned Unable to create channel of type 'zap'. I guess * is trying to open another non existent zap interface here... So, does anyone know if / how this is possible using asterisk, with just the one zap interface (x100p card)... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAP Hook flash / recall on active zap interface
I've figured out the problem my flash length was too long...
I have set flash=100 in zapata.conf
and this in extensions.conf
ignorepat=3
exten= _3XX,1,Flash()
exten= _3XX,2,SendFTMF,${EXTEN:1}
exten= _3XX,3,Flash()
exten= _3XX,4,Hangup
all works now :)
On Wed, 22 Sep 2004 11:48:24 +1000, Sophus [EMAIL PROTECTED] wrote:
Hi there,
thanks for that.
I'm using x-lite softphone, can I send a flash to the x100p from xlite
directly? or do I need to dial an extension and have asterisk do it?
any examples...?
cheers
Adam
On Thu, 16 Sep 2004 10:14:39 -0300, Marcelo Pacheco [EMAIL PROTECTED] wrote:
The flash application flashes que current ZAP device, it doesn't take a
parameter AFAIK. Correct me if I'm wrong.
For instance, you could use this for manual call deflection on a POTS line, if
the user asks to be transfered to another branch, you could use Flash then
Dial the DTMF digits to call the other extension and transfer.
Marcelo Pacheco
Em Qui 16 Set 2004 07:25, Sophus escreveu:
Hi there,
I have a x100p card in an asterisk box. Does anyone know if it's
possible to do a hook flash / recall on an active zap channel?
On what I'm trying to do...
From an ordinary analogue pstn telephone I can call someone, press
recall, call someone else, press recall 3 presto we're on a three
way chat, with me only using one line - using the telephone company's
3waychat feature...
I have tried Flash(Zap/1), and similar commands, however an error is
returned Unable to create channel of type 'zap'. I guess * is
trying to open another non existent zap interface here...
So, does anyone know if / how this is possible using asterisk, with
just the one zap interface (x100p card)...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP Hook flash / recall on active zap interface
Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 presto we're on a three way chat, with me only using one line - using the telephone company's 3waychat feature... I have tried Flash(Zap/1), and similar commands, however an error is returned Unable to create channel of type 'zap'. I guess * is trying to open another non existent zap interface here... So, does anyone know if / how this is possible using asterisk, with just the one zap interface (x100p card)... grateful for any feedback cheers sophus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
