Re: [asterisk-users] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
Hi thanks, Each span is connected to a separate Quintum gateway so I took it that each span will need to decide with its own Quintum which side is the source of the timing, I hope my logic is right because with this config I got better results. I also changed headsets to USB which improved by 200% the audio quality for both parties of the call. I am still experiencing with some settings for pci ports kernel at boot time, will update you when I finish it. Thanks again On Tue, Jan 6, 2009 at 3:28 AM, Ex Vito ex.vitor...@gmail.com wrote: IIRC, the second argument in the span lines indicates the timing sync with 1 meaning that this span is master. I'd say it makes no sense to have both of them be masters... I have no current docs / system at hand; give it a check and then, maybe try to have one as master and the other as slave (0 instead of 1, again, IIRC). Cheers, -- exvito ___ -- 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] bridge 2 calls
I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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 ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
Hi and Thanks for your reply, I am sticking to 1.2.30 because this is a Vicidial server they recommend this version. With 10 to 15 concurrent calls load on the server is 0.2 sometimes less, (thats why I was told that I don't need to recompile asterisk since I have no cpu overloading) but here is what zttest says : zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% --- Results after 15 passes --- Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586 I read on the web it should be at least 99.99% - Do you know HOW can I improve it ? maybe its the source of the problem. besides this, I paste my zaptel.conf : span=1,1,6,ccs,hdb3 span=2,1,6,ccs,hdb3 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=fr defaultzone=fr when I put 6 in LINE BUILD OUT (1,1,6,ccs,hdb3) value I got better call quality than when it was 5 (1,1,5,ccs,hdb3). I also added echotrainning = 1600 in my zapata.conf file as some website suggested. [trunkgroups] [channels] language=fr loadzone=fr defaultzone=fr context=default switchtype=euroisdn pridialplan=dynamic signalling=pri_cpe internationalprefix=+ nationalprefix=34 ;rxwink=250 ; Atlas seems to use long (250ms) winks ;rxwink=200 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotrainning=1600 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group=1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 here is interupts file : cat /proc/interrupts CPU0 CPU1 0: 84 1 IO-APIC-edge timer 1: 2 1 IO-APIC-edge i8042 8: 0 0 IO-APIC-edge rtc 9: 0 1 IO-APIC-fasteoi acpi 14: 1960 2009 IO-APIC-edge libata 15: 0 0 IO-APIC-edge libata 16: 20 21 IO-APIC-fasteoi uhci_hcd:usb1 17: 0 0 IO-APIC-fasteoi uhci_hcd:usb2 18: 41 41 IO-APIC-fasteoi libata 19: 312691 312616 IO-APIC-fasteoi wct2xxp 219: 28650 28674 PCI-MSI-edge eth0 NMI: 0 0 Non-maskable interrupts LOC: 37798 36818 Local timer interrupts RES:191216 Rescheduling interrupts CAL:153 91 function call interrupts TLB:727751 TLB shootdowns TRM: 0 0 Thermal event interrupts SPU: 0 0 Spurious interrupts ERR: 0 MIS: 0 So do you see any interrupt conflicts / sharing ? Before I go googling, Anyone knows what IO-APIC-fasteoi means ? Disk IO activity I am sure I don't have, but Ethernet activity I have a lot, the server is constantly querying a mysql database. On Mon, Jan 5, 2009 at 4:31 AM, Ex Vito ex.vitor...@gmail.com wrote: I'd start by checking for interrupt conflicts / sharing on the system. cat /proc/interrupts... also check your zap timing accuracy (was it zaptest ?) It looks as if there is some periodic IO (disk?) activity that is leading to bad audio. (also: check your logs !) :-) Then I'd add that the first impressions with the ML110 G5 were not very good regarding IO activity and the PCI bus... Not really sure what it was, as it was in the context of a few quick tests we did during the summer, but I think we had to change the BIOS defaults for the disk controller or something because we were getting really sloow IO from them. Question: Any particular reason to run 1.2 instead of something more recent ? -- exvito ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
and here is lspci output: lspci 00:00.0 Host bridge: Intel Corporation Unknown device 29f0 (rev 01) 00:01.0 PCI bridge: Intel Corporation Unknown device 29f1 (rev 01) 00:1c.0 PCI bridge: Intel Corporation Unknown device 2940 (rev 02) 00:1c.4 PCI bridge: Intel Corporation Unknown device 2948 (rev 02) 00:1c.5 PCI bridge: Intel Corporation Unknown device 294a (rev 02) 00:1d.0 USB Controller: Intel Corporation Unknown device 2934 (rev 02) 00:1d.1 USB Controller: Intel Corporation Unknown device 2935 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 92) 00:1f.0 ISA bridge: Intel Corporation Unknown device 2916 (rev 02) 00:1f.2 IDE interface: Intel Corporation Unknown device 2920 (rev 02) 00:1f.3 SMBus: Intel Corporation Unknown device 2930 (rev 02) 00:1f.5 IDE interface: Intel Corporation Unknown device 2926 (rev 02) 0d:00.0 VGA compatible controller: Matrox Graphics, Inc. MGA G200e [Pilot] ServerEngines (SEP1) (rev 02) 0e:00.0 Ethernet controller: Broadcom Corporation Unknown device 165a 11:00.0 Communication controller: Digium, Inc. Wildcard TE210P (rev 02) ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
I have a sata harddrive, do you think changing it to ata one will solve the problem ? any one has tried this solution ? ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
I am experiencing bad audio quality on my calls thru a Digium TE212P card configured for E1/euroisdn technology. My Install is : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 on a small HP Server ML110 G5 I have read I should recompile Asterisk to match my processor type (Xeon dual core), which I will be doing today. Any advice would be appreciated. Thanks in advance. ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
I forgot to describe the audio problem, well, I experience micro cuts in the voice, this does not happen during the whole call, it happens during 2 seconds then audio becomes normal, then back again 2 or 3 seconds then goes away. I have compiled libpri, zaptel asterisk with thier default config files. On Sun, Jan 4, 2009 at 8:12 AM, Nick Wolf new...@gmail.com wrote: I am experiencing bad audio quality on my calls thru a Digium TE212P card configured for E1/euroisdn technology. My Install is : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 on a small HP Server ML110 G5 I have read I should recompile Asterisk to match my processor type (Xeon dual core), which I will be doing today. Any advice would be appreciated. Thanks in advance. ___ -- 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] second trunk in extensions.conf
Sorry for being late to reply (end of the year I guess lol), I ended up
putting the two trunks under the same group dialing thru that group in my
extentions.conf
On Thu, Dec 25, 2008 at 8:27 AM, Godson Gera godso...@gmail.com wrote:
On Tue, Dec 23, 2008 at 7:32 PM, Nick Wolf new...@gmail.com wrote:
I have a TE210P digium card that has 2 E1/T1 ports.
the code in my extensions.conf file for span 1 is :
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g1; Trunk interface
TRUNKX=Zap/g2 ; 2nd trunk interface
...
...
; dial a long distance outbound number to SPAIN
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten = _00034X,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _00034X,2,Dial(${TRUNK}/${EXTEN:1},55,To)
exten = _00034X,3,Hangup
...
...
and it works fine, but I need to start working with my second span I
don't know how to add it in extensions.conf file.
you either bring both the trunks under one group in zapata.conf or replace
TRUNK with TRUNKX in your dial command, how ever that would leave first runk
idle, also you can add one more dial command as show below
exten = _00034X,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _00034X,2,Dial(${TRUNK}/${EXTEN:1},55,To)
exten = _00034X,3,Dial(${TRUNKX}/${EXTEN:1},55,To)
exten = _00034X,4,Hangup
--
Thanks Regards,
Godson Gera
http://goog_1230183919974Asterisk Consultant
Hyderabadhttp://godson.in/voip-asterisk-consultant-hyderabad-india
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[asterisk-users] second trunk in extensions.conf
I have a TE210P digium card that has 2 E1/T1 ports.
the code in my extensions.conf file for span 1 is :
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1; Trunk interface
TRUNKX=Zap/g2 ; 2nd trunk interface
...
...
; dial a long distance outbound number to SPAIN
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten = _00034X,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _00034X,2,Dial(${TRUNK}/${EXTEN:1},55,To)
exten = _00034X,3,Hangup
...
...
and it works fine, but I need to start working with my second span I don't
know how to add it in extensions.conf file.
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