[asterisk-users] Codec and CPU load
Hi as maximum link capacity could be calculated using codecs and channel types so , regarding the CPU and processors load , Is there any formula or (any relations could help ) that can give the maximum CPU load (mainly processor and RAM ) or scalability average using asterisk channels , codecs , applications …. Ayman _ See what people are saying about Windows Live. Check out featured posts. http://www.windowslive.com/connect?ocid=TXT_TAGLM_WL_connect2_082008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HP server and Meetme applications
Hi list I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ? and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? AyMaN ALMONTAHA .ICT 11 AUG 2008 _ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Langugae issue
Hi list I add new directory for Arabic voices support and I 'd translated all the English voices files into Arabic , with language = ar ,and it is working fine ,except some problems in saying the number order ,because the Arabic structure is quite different for numbers ,where in English language we can say twenty two while the order should be two and twenty ,so please if you can guide me how to change the setting to do that . regads Ayman _ Watch “Cause Effect,” a show about real people making a real difference. Learn more. http://im.live.com/Messenger/IM/MTV/?source=text_watchcause___ -- 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] Asterisk and Digium HP thin clients compatibility
Hi all i hve check the HP thin clinets web site http://h10010.www1.hp.com/wwpc/us/en/sm/WF04a/12454-12454-321959-338927-89307.html and i found that they hve debian and NeoLinux OS with AMD Sempron 2100+ and AMD Geode™ NX 1500 1.0 GHz processor ,so is asterisk , digium analog cards, and zaptel driver can be installed and working fine for these two types of processor for debian and Neolinux platform regards ayman _ In a rush? Get real-time answers with Windows Live Messenger. http://www.windowslive.com/messenger/overview.html?ocid=TXT_TAGLM_WL_Refresh_realtime_042008___ -- 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] [ [asterisk-ss7] libss7 2asterisk box
Hi list
I plan to connect two asterisk box using libss7 ,i read the list messages (
thanks for this great jop) , i installed all the packegs with digium single E1
link in both boxes with cenos 5 and every thing is looking ok excact when i am
trying to call using sip channel it shows some problems here is muy
configrations file
server A--B
zaptel.conf
span=1,0,0,ccs,hdb3 ;span=1,1,0,ccs,hdb3 server B
bchan=1-15,17-31
dchan=16
loadzone = us
defaultzone = us
ztcfg -vv
Zaptel Version: SVN--rEcho Canceller: MG2Configuration==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)Channel 02: Clear channel
(Default) (Slaves: 02)Channel 03: Clear channel (Default) (Slaves: 03)Channel
04: Clear channel (Default) (Slaves: 04)Channel 05: Clear channel (Default)
(Slaves: 05)Channel 06: Clear channel (Default) (Slaves: 06)Channel 07: Clear
channel (Default) (Slaves: 07)Channel 08: Clear channel (Default) (Slaves:
08)Channel 09: Clear channel (Default) (Slaves: 09)Channel 10: Clear channel
(Default) (Slaves: 10)Channel 11: Clear channel (Default) (Slaves: 11)Channel
12: Clear channel (Default) (Slaves: 12)Channel 13: Clear channel (Default)
(Slaves: 13)Channel 14: Clear channel (Default) (Slaves: 14)Channel 15: Clear
channel (Default) (Slaves: 15)Channel 16: D-channel (Default) (Slaves:
16)Channel 17: Clear channel (Default) (Slaves: 17)Channel 18: Clear channel
(Default) (Slaves: 18)Channel 19: Clear channel (Default) (Slaves: 19)Channel
20: Clear channel (Default) (Slaves: 20)Channel 21: Clear channel (Default)
(Slaves: 21)Channel 22: Clear channel (Default) (Slaves: 22)Channel 23: Clear
channel (Default) (Slaves: 23)Channel 24: Clear channel (Default) (Slaves:
24)Channel 25: Clear channel (Default) (Slaves: 25)Channel 26: Clear channel
(Default) (Slaves: 26)Channel 27: Clear channel (Default) (Slaves: 27)Channel
28: Clear channel (Default) (Slaves: 28)Channel 29: Clear channel (Default)
(Slaves: 29)Channel 30: Clear channel (Default) (Slaves: 30)Channel 31: Clear
channel (Default) (Slaves: 31)
31 channels to configure.
zapata.conf
[trunkgroups]
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
; Options for use with signalling=ss7 -
signalling=ss7
ss7type = itu
;ss7_called_nai=dynamic
linkset = 1
pointcode =5770 ; 5760 server B
adjpointcode = 5760 ;5770 server B
defaultdpc = 5760 ;5770 server B
networkindicator=national
context=ss7
sigchan = 16
cicbeginswith=1
channel=1-15
cicbeginswith=17
channel=17-31
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
[default]
exten = s,1,Answer()
exten = s,2,Playback(hello-world)
exten = s,3,hangup()
include =ss7
include =123
[ss7]
exten = s,1,Answer()
exten = s,2,Playback(hello-world)
exten = s,3,hangup()
[123]
include =ss7
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _,1,Dial(Zap/r1/${EXTEN})
when do cli asterisk at server A
Asterisk Ready. == Parsing '/etc/asterisk/cli.conf': == Found*CLI --- SS7
Up ---Resetting CICs 1 to 15Resetting CICs 17 to 31Got reset acknowledgement
from CIC 1 to 15.Got reset acknowledgement from CIC 17 to 31.
= Using SIP RTP CoS mark 5-- Executing [EMAIL PROTECTED]:1]
Dial(SIP/105-099c4e80, Zap/r1/1105) in new stack-- Called r1/1105
WARNING[3689]: app_dial.c:824 wait_for_answer: Unable to forward voice or
dtmfWARNING[3689]: app_dial.c:824 wait_for_answer: Unable to forward voice or
dtmf -- Hungup 'Zap/1-1' -- No one is available to answer at this time
(1:0/0/0) -- Auto fallthrough, channel 'SIP/105-099c4e80' status is 'NOANSWER'
server B
NOTICE[4160]: chan_zap.c:9696 ss7_linkset: Received RLC out and we haven't
sent REL. Ignoring.
thanx in advance
ayman
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Re: [asterisk-users] real zaptel call durations
Thanx alot for reply I mean i have to use the fxo to connect to the pstn line and i do not know if there is any asterisk functions ,Application, options that could help to know what is the real call duration [ how to deal with pstn line signaling how to detect the pstn ringing tone or pstn auto-machinese voice message in case if the user did not answer the call ] , i saw some billing software and i am not sure if they are calculating the bills using cdr in case of using fxo. thank u in advance ayman _ Helping your favorite cause is as easy as instant messaging. You IM, we give. http://im.live.com/Messenger/IM/Home/?source=text_hotmail_join___ -- 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] real zaptel call durations
How to calculate the PSTN call durations through zaptel ,where in the CDR it gives the time durations started when the zaptel answerd + PSTN dialing time + ringing time even thoug the destinations did not answer the call , so as a reult i will find user X is dialing PSNT line for 40 seconds even though the distination did not answer the call. _ Need to know the score, the latest news, or you need your Hotmail®-get your fix. http://www.msnmobilefix.com/Default.aspx___ -- 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] mfcr2 stuck
Hi Jakub could you please post the zaptel.conf and asterisk cli unicall channel error plus what is version of unicall and zaptel are u install and i think you miss the protocolend= option (cpe or net??) line in the uncall.conf ayman From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Mon, 18 Feb 2008 08:36:52 +0100Subject: [asterisk-users] mfcr2 stuck Hello I'm using mfcr2 support (unicall) in asterisk 1.4. Everything is working fine, asterisk can answer calls. But after some random period of time mfcr2 module stuck. When I make a call to my * box I can hear only signal of getting caller ID („tritirirti” – like jumping on :) ) and connection is terminated by my telecom operator. When everything I ok after few seconds of this signal my asterisk answer. Below is my unciall.conf [EMAIL PROTECTED] ~]# cat /etc/asterisk/unicall.conf [Channels] loglevel=255 language=pl context=from-pstn usecallerid=no hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=800 echotraining=yes relaxdtmf=no rxgain=0 txgain=0 group=11 callgroup=0 pickupgroup=0 immediate=yes callerid=asreceived amaflags=default accountcode=avantel musiconhold=default protocolclass=mfcr2 protocolvariant=cz,9,6 channel=1-15 channel=17-31 supertones=pl Regards Jakub _ Shed those extra pounds with MSN and The Biggest Loser! http://biggestloser.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
Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports
Hi Bilal could you post the TDM configuration file (zaptel.conf and zapata.conf) and how did you compile it Regards Ayman Date: Wed, 13 Feb 2008 04:35:43 -0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports Hi All; I am facing a problem that the telephon line in Egypt does not work with the FXO port at the digium card (TDM22B), and I tried to play in loadzone and defaultzone without any success, when we call to the PBX it gives Busy signal sometimes, and othertimes it rings without any response in Asterisk. Is there any other configuration I have to do it to resolve this issue? Any advise about a troubleshooting method to resolve it? Any help? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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 _ Helping your favorite cause is as easy as instant messaging. You IM, we give. http://im.live.com/Messenger/IM/Home/?source=text_hotmail_join___ -- 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
