[asterisk-users] Needs more cpu usage
Hi, I am running * on centos5 using 4core cpu. When it is busy, * uses 99.9% of cpu max. How can I make * to use more cpu power? Thanks. ___ -- 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] CPU Usage
Hi, I'm runnung * on centos4 smp. When system is busy, asterisk uses 99.9% cpu. I want asterisk to use more 100% cpu to process more calls. Is this possible? Thanks. ___ -- 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] Tone detection while Dialing
Hi, I want to detect a tone while Dial() through pri. When a secial tone(eg, #), I want to send the call to another extenison. Regards. Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --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] Call transfer while dialing
Hi, I want to transfer the call to a conferencing room while dialing. I tried to do that using manager API(Redirect), but it did't work. Regards, Jason. Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ipv6 patch
Is it exists? Regards, Hong Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support DNIS?
Would you attach your whole zaptel.conf and zapata.conf? --- C F [EMAIL PROTECTED] wrote: Also check out immediate=no On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: David Ruggles wrote: I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. We had to add this to the /etc/asterisk/zapata.conf to make Asterisk work with the EM Wink start T-1 from our telco. I guess I could paste the settings this time. wink=270 rxwink=270 You might want to play with those settings. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spliting video and audio
Hi, This is the configuration I want. Hard Video phone---video---Soft Video Phone(PC) ^ | audio | V Audio Only Phone Any idea? Regards, Jason Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] snom 360 auto answer
Thankyou David,
It works for Linksys,but not for snom 360.
Do I need to change someting using web UI ?
--- Klaverstyn, David C
[EMAIL PROTECTED] wrote:
This is my code (that I copied form somewhere) for
paging a group of
phones. By dialling 99 it will page phones 2101,
2102 and 2105.
Just include the context ext-paging in your dial
plan and modify the
extension numbers and all should be good.
This works on Linksys Phones but should also work on
Snoms.
I hope this helps you.
[ext-paging]
exten = PAGE2101,1,GotoIf($[ ${CALLERID(number)} =
2101 ]?skipself)
exten = PAGE2101,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2101,n,Dial(SIP/2101,5)
exten = PAGE2101,n(skipself),Noop(Not paging
originator)
exten = PAGE2102,1,GotoIf($[ ${CALLERID(number)} =
2102 ]?skipself)
exten = PAGE2102,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2102,n,Dial(SIP/2102,5)
exten = PAGE2102,n(skipself),Noop(Not paging
originator)
exten = PAGE2105,1,GotoIf($[ ${CALLERID(number)} =
2105 ]?skipself)
exten = PAGE2105,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2105,n,Dial(SIP/2105,5)
exten = PAGE2105,n(skipself),Noop(Not paging
originator)
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]
aging)
exten =
99,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${EXTEN:2})
exten = _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = _99,n,Set(__ALERT_INFO=Ring Answer)
exten = _99,n,Dial(SIP/${EXTEN:2})
__
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[asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${EXTEN:2})
exten = _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = _99,n,Set(__ALERT_INFO=Ring Answer)
exten = _99,n,Dial(SIP/${EXTEN:2})
__
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[asterisk-users] chan_oh323 early media
Hi, I configured openh323_v1_18_0, pwlib_v1_10_0 and asterisk-oh323-0.7.3. I can call inbound and outbound. But early media is not working in outboubd. Regards, Jason. oh323.conf == [general] listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 ;fastStart=yes fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=yes jitterMin=20 jitterMax=500 ipTos=reliability outboundMax=20 inboundMax=20 ;bandwidthLimit=1024 wrapLibTraceLevel=10 libTraceLevel=10 ;wrapLibTraceLevel=0 ;libTraceLevel=0 libTraceFile=stdout gatekeeper=192.168.1.150 gatekeeperTTL=60 ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 userInputMode=TONE amaFlags=billing accountCode=aaabbbaaabbb context=from-323 [register] context=from-323 alias=MyH323ID alias=555 alias=5556667 alias=5556668 [codecs] codec=G711A frames=20 codec=G711U frames=20 codec=G7231 frames=20 codec=G729A frames=20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H323 NAT Problem
Hi, I installed asterisk with oh323. My gatekeeper is out of nat device. How can i register * to gatekeeper? Thanks in advance.. Jason. Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4 chan_h323, help please...
It' seems to be RTP problem. sendto() in rtp.c fails to send rtp packets. When I change channel from H323 to SIP, no problem. Any idea? Regards, Jason. -- from /var/log/asterisk/full-- [Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission error of packet 39407 to 192.168.1.116:8528: Invalid argument [Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission error of packet 39408 to 192.168.1.116:8528: Invalid argument [Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission error of packet 39409 to 192.168.1.116:8528: Invalid argument --- Jason Kim [EMAIL PROTECTED] wrote: Hi, My configuration is SipPhone--*1---*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)--asterisk1---(H323)--asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best h323 channel driver? Regards, Jason. #--h323.conf for both [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw context=default #--dial plan of asterisk1 exten = *59,1,Wait(1) exten = *59,2,Dial(H323/[EMAIL PROTECTED]) #--dial plan of asterisk2 exten = 3500,1,Playback(hello) exten = 3500,2,Hangup() #--console messages with 'rtp debug'- -- Executing [EMAIL PROTECTED]:3] Dial(SIP/3503-0921cb88, H323/[EMAIL PROTECTED]) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper. == New H.323 Connection created. -- root is calling host [EMAIL PROTECTED]:1720 -- Call token is ip$localhost/29426 -- Call reference is 29426 -- DTMF Payload is [pt=101] -- Called [EMAIL PROTECTED] Setting capabilities to 0x8 (alaw) Capabilities in preference order is (alaw) Allowed Codecs: Table: G.711-ALaw-64k 1 UserInput/hookflash 2 UserInput/RFC2833 3 UserInput/dtmf 4 Set: 0: 0: G.711-ALaw-64k 1 1: UserInput/hookflash 2 2: UserInput/RFC2833 3 UserInput/dtmf 4 -- Sending SETUP message -- Transmitting RFC2833 on payload 101 -- Started logical channel: receiving G.711-ALaw-64k -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 127.0.0.1 -- remotePort: 13710 -- ExternalIpAddress: 192.168.1.116 -- ExternalPort: 29388 -- Started logical channel: sending G.711-ALaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 127.0.0.1 -- remotePort: 13710 -- ExternalIpAddress: 192.168.1.116 -- ExternalPort: 29388 - Progress Indicator: 8 -- H323/192.168.1.150-3 is making progress passing it to SIP/3503-0921cb88 -- Inbound RFC2833 on payload [pt=101] Peer capability is G.711-ALaw-64k 1 Found peer capability G.711-ALaw-64k 1, Asterisk code is 8, frame size (in ms) is 20 Peer capability is UserInput/hookflash 2 Peer capability is UserInput/RFC2833 3 Peer capability is UserInput/dtmf 4 Peer capabilities = 0x8 (alaw), ordered list is (alaw) =-= In OnConnectionEstablished for call 29426 -- Connection Established with 3500 -- H323/192.168.1.150-3 answered SIP/3503-0921cb88 -- Received Facility message... Got RTP packet from192.168.1.204:16434 (type 00, seq 014405, ts 328224084, len 000240) Sent RTP packet to 127.0.0.1:13710 (type 08, seq 008392, ts 96, len 000160) Got RTP packet from192.168.1.204:16434 (type 00, seq 014406, ts 328224324, len 000240) Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 chan_h323, help please...
Hi, My configuration is SipPhone--*1---*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)--asterisk1---(H323)--asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best h323 channel driver? Regards, Jason. #--h323.conf for both [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw context=default #--dial plan of asterisk1 exten = *59,1,Wait(1) exten = *59,2,Dial(H323/[EMAIL PROTECTED]) #--dial plan of asterisk2 exten = 3500,1,Playback(hello) exten = 3500,2,Hangup() #--console messages with 'rtp debug'- -- Executing [EMAIL PROTECTED]:3] Dial(SIP/3503-0921cb88, H323/[EMAIL PROTECTED]) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper. == New H.323 Connection created. -- root is calling host [EMAIL PROTECTED]:1720 -- Call token is ip$localhost/29426 -- Call reference is 29426 -- DTMF Payload is [pt=101] -- Called [EMAIL PROTECTED] Setting capabilities to 0x8 (alaw) Capabilities in preference order is (alaw) Allowed Codecs: Table: G.711-ALaw-64k 1 UserInput/hookflash 2 UserInput/RFC2833 3 UserInput/dtmf 4 Set: 0: 0: G.711-ALaw-64k 1 1: UserInput/hookflash 2 2: UserInput/RFC2833 3 UserInput/dtmf 4 -- Sending SETUP message -- Transmitting RFC2833 on payload 101 -- Started logical channel: receiving G.711-ALaw-64k -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 127.0.0.1 -- remotePort: 13710 -- ExternalIpAddress: 192.168.1.116 -- ExternalPort: 29388 -- Started logical channel: sending G.711-ALaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 127.0.0.1 -- remotePort: 13710 -- ExternalIpAddress: 192.168.1.116 -- ExternalPort: 29388 - Progress Indicator: 8 -- H323/192.168.1.150-3 is making progress passing it to SIP/3503-0921cb88 -- Inbound RFC2833 on payload [pt=101] Peer capability is G.711-ALaw-64k 1 Found peer capability G.711-ALaw-64k 1, Asterisk code is 8, frame size (in ms) is 20 Peer capability is UserInput/hookflash 2 Peer capability is UserInput/RFC2833 3 Peer capability is UserInput/dtmf 4 Peer capabilities = 0x8 (alaw), ordered list is (alaw) =-= In OnConnectionEstablished for call 29426 -- Connection Established with 3500 -- H323/192.168.1.150-3 answered SIP/3503-0921cb88 -- Received Facility message... Got RTP packet from192.168.1.204:16434 (type 00, seq 014405, ts 328224084, len 000240) Sent RTP packet to 127.0.0.1:13710 (type 08, seq 008392, ts 96, len 000160) Got RTP packet from192.168.1.204:16434 (type 00, seq 014406, ts 328224324, len 000240) Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H323 no audio
Hi, My configuration is SipPhone-asterisk1 -asterisk2. My asterisk version is 1.2.10. I installed chan_h323 according to 'http://astrecipes.net/?n=102'. When i call from asterisk1 to asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Regards, Jason. #--h323.conf for both [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw context=default #--dial plan of asterisk1 exten = *59,1,Wait(1) exten = *59,2,Dial(H323/[EMAIL PROTECTED]) #--dial plan of asterisk2 exten = 3500,1,Playback(hello) exten = 3500,2,Hangup() #--'rtp debug' messages-- Raw PDU: 08 02 55 13 62 1c 00 7e 00 0f 05 28 10 01 00 04 ..U.b..~...( c0 01 80 05 01 03 28 00 01 ..(.. 2:15:36.845 H225 Caller:89bf340 h323.cxx(4301) H323 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged Got RTP packet from 192.168.1.232:16426 (type 0, seq 1540, ts 161645797, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1541, ts 161646037, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1542, ts 161646277, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1543, ts 161646517, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1544, ts 161646757, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1545, ts 161646997, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1546, ts 161647237, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1547, ts 161647477, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1548, ts 161647717, len 240) Sponsored Link Don't quit your job - take classes online www.Classesusa.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-114 noise
Jessee, Thank you for your help. I downloaded firmware and sample configuration files. But the firmware was old version for MP118 and MP124. Where can i download recent one? Can i upload only ini file to change countrycoefficient ? Regards, Jason. --- Jessee J Holmes [EMAIL PROTECTED] wrote: Jason, First, before you start reading, get to the latest firmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), there have been significant echo improvements in this version. After many days of working with Audiocodes on this problem and much time spent here by multiple technicians trying to reproduce and resolve this issue; this morning, Atacomm received an email from Audiocodes with a full explanation to this now confirmed issue with all MP-11x units. Atacomm will immediately begin work on a KB article within our website that confirms this issue and outlines the manufacturer recommended steps to resolve this problem. Apparently, there have been some changes with the MP-11x's that can negatively affect line noise and echo. Below are some steps which can help to correct these problems: 1. The new design did away with the Coefficent file. Audiocodes, now instead, introduced a configurable parameter called countrycoefficient. This parameter can be adjusted to a specific country based on known configurations. For the most part this should work. 70(USA) is the default value. More can be found in the Users manual. 2. In just about every case, an FXO is added to a Pre-existing PBX or CO line, you can expect echo. This comes from the fact that delay (IP Network) is being introduced, and what used to be Side tone is now delayed so much it is echo. Just about every difference on the line that can be heard between the pre fxo and post fxo installation can be traced to echo, or line quality issues. 3. Going forward, Audiocodes would like to suggest that when installing the product do the following: A) Make sure the Line coming from the PBX or CO is a Loop Start line. Ground start is not supported on the MP-11x series of gateways. (The M1K FXO will in 5.0) B) Check that the Line can deliver for a 600 Ohm Impedance line -52 to -24 V of Off Hook Voltage -15 to -6 V of On Hook Voltage 20 to 35 ma of loop current. If you know the line is not 600 Ohm, please gather metrics on the line, and the make and model of the PBX or switch it is attached too, plus country of origin. If it is not from the USA, please look up the country of origin and then find the CountryCoefficient to match this. Load the .ini file to the board with this setting and reset. Make sure the Gateway has a firmware version of 4.60.035 or higher or 4.80.030 or higher. C) Put the device on the network with Voice Volume set to 0 and input gain set to 0. Make calls, if there is no issue, you can stop here. However, Echo is still expected most of the time. D) The echo should be heard by the IP side participant as their voice is reflected back. If this is the case, then what needs to be done is to lower the voicevolume (IPTEL). This way the speakers reflected voice will comeback low enough for the ECAN to cancel it out (-6 is usually recommended as the value to plug in here). A little experimentation is needed as the loss for all lines will vary based on length from the CO. Echo is usually taken care of in this manner. E) The incoming speaker from the PSTNs voice seems low, set InputGainLocation =1, and then slowly increment the Input Gain Parameter(Tel?P) to adjust for this. In past releases (see the note about loads above), the input gain was always applied prior to the ECAN which had the effect of amplifying the returned echo and noise on the line causing crosstalk and clipping issues. This is no longer the case. If the above does not resolve the issues, then you need to go ahead and collect DSP, Ethereal and Syslog traces along with the board.ini, these are to be sent to your support agent, who will then send these to Audiocodes for their engineers to evaluate. This should not happen often. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Nov 3, 2006, at 12:14 AM, Jason Kim wrote: Jessee, I tried many combinations of Voice Volume, Input Gain and packetization time , but it's noisy steel. I'm using G.711A-law and packetization time is 20ms. It can be impedance mismatch problem but i cannot adjust impedance of FXO port of MP-114. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re: [asterisk-users] Audiocodes MP-114 noise
Jessee, I tried many combinations of Voice Volume, Input Gain and packetization time , but it's noisy steel. I'm using G.711A-law and packetization time is 20ms. It can be impedance mismatch problem but i cannot adjust impedance of FXO port of MP-114. --- Jessee J Holmes [EMAIL PROTECTED] wrote: Jason, There are a couple things we can try to fix your problem. Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmp Let's try some quick things first though: In your web interface, go to advanced config - channel settings / voice settings There are some options here you can play with: Voice Volume (IP side of this thing) - by default this should be set at '1'. Try bringing this down slowly, I'd say in increments of 5 (-4, then -9, and so on). Range on this option can be anywhere from -32 to +32, you really shouldn't need to go beyond -15; but you're actual volume on the calls should still stay reasonable. Input Gain (telco side) is another option you can slowly change as well (set to 0 by default). There should also be spot where you can specify the codername, you could possibly try changing this to another codec such as G.729 or G. 711u-law (should be the same codec being used on your Asterisk system) try changing packet size from 20 to 40 or 60. This may also help. If none of this stuff helps, let me know. We can then start getting really technical. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Nov 1, 2006, at 1:45 AM, Jason Kim wrote: Thank you Jessee, Firmware seems to be recent(4.80A.025.004). For 'noisy', I mean IP Phone -- * -- MP-114 side. Audio quality of MP-114 -- PSTN -- Analog phone is good. I think it can be power ground or gain problem. Any experience? Thanks, Jason --- Jessee J Holmes [EMAIL PROTECTED] wrote: Dear Jason, Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well? This start happening immediately, or was the box working before and the problem just started? Also, a quick heads up, make sure before even beginning to troubleshoot an issue like this you do a factory reset to the unit and get the latest available firmware on it. Usually that fixes annoying issues like this. Thanks, Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Oct 30, 2006, at 10:36 PM, Jason Kim wrote: It's noisy while talking. Any idea? Thanks in advance. Jason __ __ Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ __ Low, Low, Low Rates! Check out Yahoo! Messenger's cheap PC-to-Phone call rates (http://voice.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Check out the New Yahoo! Mail - Fire up a more powerful email and get things done faster. (http://advision.webevents.yahoo.com/mailbeta) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-114 noise
Thank you Jessee, Firmware seems to be recent(4.80A.025.004). For 'noisy', I mean IP Phone -- * -- MP-114 side. Audio quality of MP-114 -- PSTN -- Analog phone is good. I think it can be power ground or gain problem. Any experience? Thanks, Jason --- Jessee J Holmes [EMAIL PROTECTED] wrote: Dear Jason, Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well? This start happening immediately, or was the box working before and the problem just started? Also, a quick heads up, make sure before even beginning to troubleshoot an issue like this you do a factory reset to the unit and get the latest available firmware on it. Usually that fixes annoying issues like this. Thanks, Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Oct 30, 2006, at 10:36 PM, Jason Kim wrote: It's noisy while talking. Any idea? Thanks in advance. Jason __ __ Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Low, Low, Low Rates! Check out Yahoo! Messenger's cheap PC-to-Phone call rates (http://voice.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes MP-114 noise
It's noisy while talking. Any idea? Thanks in advance. Jason Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handytone 286 T.38 SDP parameters
Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is FaxMachine1--HandyTone286--asterisk--spa2100--FaxMachine2. When I send fax from FaxMachine1, I cannot see any T.38 SDP parameters. Any idea is appreciated. Thanks. Jason. [general] bindport=5060 bindaddr=0.0.0.0 allow=all t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF relay
Hi, My environment is ITSP---Asterisk--SipPhone. I want to send dtmf from SipPhone to ISTP using 'info' or 'rfc2833'. Is this possible? Thanks. Jason. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pri Gateway Hardware
Does TDMoE supports kernel 2.6? Where should I do echo cancellation? --- Carlos Alperin [EMAIL PROTECTED] wrote: Low level requeriment, just you transfer everything using level 2. So you don't need to the overhead to have Asterisk running to route that traffic. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, January 10, 2006 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pri Gateway Hardware Alexander Lopez a ?rit : TDMoE is stable and stale, There is no more development planed or needed as it only opens up a pipe between two ethernet points using Layer 2. OK... What would be in the advantage in running TDMoE rather than using IAX or SIP? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Remote Call Control
Hi All, I want to control a sip phone from my pc. I found a draft for this. http://www.faqs.org/ftp/pub/internet-drafts/draft-mahy-sip-remote-cc-01.txt Can someone let me know sip phones supporting this protocol or similar one? Thanks. Jason. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HW Echo Cancellers
Hi, To solve echo problems, I'm considering 2 alternatives. 1 Sangoma A104d - I can't find support for asterisk 1.2.1 2 Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command reload can change rx_gain and tx_gain? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sangoma a104d install
Hi, While a104d install on asterisk 1.2 and CVS-HEAD patch for zaptel.c failed. Is it avaiable not yet? Thanks. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] sangoma a104d
Hi, TE406p seems to be unstable yet. I ordered an sangoma a104d. Does anyone have experience of this card? Thanks. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Chanisavail and IAX2
Hi, Im trying to do this: exten = s,7,ChanIsAvail(IAX2/agent) I searched google and found that on cvs-head ChanisAvail(IAX2) is not working. I need both cvs-head and ChanisAvail. Any idea? Thanks. http://lists.digium.com/pipermail/asterisk-users/2005-March/096682.html __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Queue application problem
Hi, I've posted this problem before, but no response. I'm using iaxcomm for agents, and sometimes when there are agents waitng, incomming calls are not connected to agents for 20~30 seconds. In that case one agent is displayed in Ringing state. How can i avoid this situation? Any response is highly appreciated. Thanks. queue.conf -- [general] ;monitor-format = gsm [default] timeout = 4 maxlen = 0 music=default [que1] leavewhenempty=no music=default strategy=leastrecent joinempty=yes eventwhencalled=yes retry=1 CLI shoq eueues -- que1 has 12 calls (max unlimited) in 'leastrecent' strategy (32s holdtime), W:0, C:883, A:411, SL:0.0% within 0s Members: IAX2/agent05 (dynamic) (Not in use) has taken no calls yet IAX2/agent11 (dynamic) (Not in use) has taken no calls yet IAX2/agent23 (dynamic) (Not in use) has taken no calls yet IAX2/agent16 (dynamic) (Not in use) has taken no calls yet IAX2/agent09 (dynamic) (Not in use) has taken no calls yet IAX2/agent06 (dynamic) (Not in use) has taken no calls yet IAX2/agent12 (dynamic) (Not in use) has taken 1 calls (last was 44 secs ago) IAX2/agent15 (dynamic) (Ringing) has taken no calls yet Callers: 1. Zap/38-1 (wait: 1:32, prio: 1) 2. Zap/49-1 (wait: 0:51, prio: 1) 3. Zap/51-1 (wait: 0:47, prio: 1) 4. Zap/52-1 (wait: 0:40, prio: 1) 5. Zap/39-1 (wait: 0:28, prio: 1) 6. Zap/41-1 (wait: 0:21, prio: 1) 7. Zap/53-1 (wait: 0:19, prio: 1) 8. Zap/54-1 (wait: 0:16, prio: 1) 9. Zap/43-1 (wait: 0:05, prio: 1) 10. Zap/55-1 (wait: 0:05, prio: 1) 11. Zap/44-1 (wait: 0:04, prio: 1) 12. Zap/58-1 (wait: 0:04, prio: 1) iax.conf -- [general] port=5036 disallow=all allow=alaw jitterbuffer=yes maxjitterbuffer=300 maxexccessbuffer=50 tos=0x04 qualify=no [agent00] type=friend username=agent00 secret=agent00 context=agent host=dynamic notransfer=yes __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] one way audio
Hi, My configuration is iaxcomm--*(Sangoma 102)--pri. About 5% of iaxcomm---pri calls loose iaxcomm--pri direction audio during conversation. Any idea? Thanks. iax.conf --- [general] port=5036 disallow=all allow=alaw jitterbuffer=yes maxjitterbuffer=500 maxexccessbuffer=50 tos=0x04 qualify=no [agent1] type=friend username=agent1 secret=agent1 context=agent host=dynamic notransfer=yes zapata.conf --- [channels] context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes callprogressdetect=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=4.0 txgain=-2.0 group=1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Queue delay
Hi, * queue application delays about 10 seconds to connect to the agents. queue.conf - [general] ;monitor-format = gsm [default] timeout = 4 ; How long do we wait before trying all the members again? retry=1 ; Maximum number of people waiting in the queue (0 for unlimited) maxlen = 0 music=default [que1] leavewhenempty=no music=default strategy=leastrecent joinempty=yes CLI show queues que1 has 3 calls (max unlimited) in 'leastrecent' strategy (25s holdtime), W:0, C:189, A:29, SL:0.0% within 0s Members: IAX2/agent7 (dynamic) (Not in use) has taken no calls yet IAX2/agent6 (dynamic) (Not in use) has taken no calls yet IAX2/agent8 (dynamic) (Not in use) has taken no calls yet IAX2/agent3 (dynamic) (Ringing) has taken no calls yet Callers: 1. Zap/48-1 (wait: 0:21, prio: 1) 2. Zap/46-1 (wait: 0:18, prio: 1) 3. Zap/49-1 (wait: 0:00, prio: 1) How can i reduce connection time? Thanks, Jason __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406p install?
Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Is digium supporting new te405p and te406p install?
I tried both 1.0.9 and 1.2beta. I couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. --- Jason Walker [EMAIL PROTECTED] wrote: I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM server just waiting and try it every so often. When I am using a card for timing (TE405P is what we pretty much use), I feel pretty comfortable with FC1 and 1.0.9. Are you using 1.0.9? Have you tried 1.2 beta? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 7:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is digium supporting new te405p and te406p install? Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Is digium supporting new te405p and te406pinstall?
I was happy with FC3, old te405p and * 1.0.7. I've been thinking that kernel 2.6 is more stable and secure. --- Jason Walker [EMAIL PROTECTED] wrote: I kept running into compile errors when dealing with my Compaq (it is an older quad 700 Xeon...not sure of the model number). Once I dropped to FC1, the install of 1.0.9 compiled and install without an issue. Is there some other process/app that you are running that requires the newer kernel? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 8:54 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Is digium supporting new te405p and te406pinstall? I tried both 1.0.9 and 1.2beta. I couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. --- Jason Walker [EMAIL PROTECTED] wrote: I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM server just waiting and try it every so often. When I am using a card for timing (TE405P is what we pretty much use), I feel pretty comfortable with FC1 and 1.0.9. Are you using 1.0.9? Have you tried 1.2 beta? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 7:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is digium supporting new te405p and te406p install? Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TE406p no interrupts
Hi, I've installed an TE406p, asterisk1.2 on tyan opteron board. After installation there is no interrupts from TE406p. Is this board stable? Should i change * version to 1.0.9? Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] TE406p no interrupts
zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 cat /proc/interrupts -- CPU0 CPU1 0: 200570 273687IO-APIC-edge timer 4: 0 16IO-APIC-edge serial 8: 0 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 169: 1425 3343 IO-APIC-level libata, ehci_hcd, ohci_hcd, ohci_hcd 177: 0 2 IO-APIC-level AMD AMD8111, uhci_hcd, ohci1394 185: 1979 30 IO-APIC-level uhci_hcd, eth0 193: 4 4 IO-APIC-level wct4xxp NMI: 18 22 LOC: 474031 474031 ERR: 0 MIS: 0 Thanks. --- Alexander Lopez [EMAIL PROTECTED] wrote: Did it take an interrupt?? Whats does /proc/interrupts say?? Did you check your span= settings in zaptel.conf?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Sunday, September 11, 2005 5:48 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE406p no interrupts Hi, I've installed an TE406p, asterisk1.2 on tyan opteron board. After installation there is no interrupts from TE406p. Is this board stable? Should i change * version to 1.0.9? Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! for Good Watch the Hurricane Katrina Shelter From The Storm concert http://advision.webevents.yahoo.com/shelter ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] TE406p no interrupts
I'm using FC3. uname -a - Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2 15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 cat /proc/interrupts -- CPU0 CPU1 0: 200570 273687IO-APIC-edge timer 4: 0 16IO-APIC-edge serial 8: 0 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 169: 1425 3343 IO-APIC-level libata, ehci_hcd, ohci_hcd, ohci_hcd 177: 0 2 IO-APIC-level AMD AMD8111, uhci_hcd, ohci1394 185: 1979 30 IO-APIC-level uhci_hcd, eth0 193: 4 4 IO-APIC-level wct4xxp NMI: 18 22 LOC: 474031 474031 ERR: 0 MIS: 0 Thanks. --- Boris Bakchiev [EMAIL PROTECTED] wrote: What kernel are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Sunday, September 11, 2005 7:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE406p no interrupts Hi, I've installed an TE406p, asterisk1.2 on tyan opteron board. After installation there is no interrupts from TE406p. Is this board stable? Should i change * version to 1.0.9? Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] TE406p no interrupts
I modified wct4xxp.c and make clean; make linux26; make install; reboot; But the system is not rebooted. Because the system is in remote office I will check it next morning. Could you let me know your linux version, * version and motherboard? Thank you Boris. --- Boris Bakchiev [EMAIL PROTECTED] wrote: Well. Try this please (but only if you're running on the latest sources). Open wct4xxp.c sources and search for pci_module_init Replace it with pci_register_driver So the line should read: res = pci_register_driver(t4_driver); That allows you to get the card working on 2.6.13 in almost exactly the same setup as yours. One weird thing though. Do no use insmod ./wct4xxp.ko from zaptel directory as it will not work. Do a proper make install and then modprobe. This is just part of the fixes you might need to do. If you encounter a problem after span reconfiguration (ztcfg) let me know. If you get stuck.. let me know. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Sunday, September 11, 2005 8:14 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] TE406p no interrupts I'm using FC3. uname -a - Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2 15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] TE406p no interrupts
I modified wct4xxp.c and installed it. This is the message for 'modprobe wct4xxp' -- FATAL: Error inserting wct4xxp (/lib/modules/2.6.9-1.667smp/extra/wct4xxp.ko): No such device FATAL: Error running install command for wct4xxp astpbx kernel: Oops: [1] SMP astpbx kernel: CR2: a0362081 Regards, Jason --- Boris Bakchiev [EMAIL PROTECTED] wrote: You should have just done this: rmmod wct4xxp rmmod zaptel modprobe wct4xxp It will do the same thing -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Monday, 12 September 2005 00:34 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] TE406p no interrupts I modified wct4xxp.c and make clean; make linux26; make install; reboot; But the system is not rebooted. Because the system is in remote office I will check it next morning. Could you let me know your linux version, * version and motherboard? Thank you Boris. --- Boris Bakchiev [EMAIL PROTECTED] wrote: Well. Try this please (but only if you're running on the latest sources). Open wct4xxp.c sources and search for pci_module_init Replace it with pci_register_driver So the line should read: res = pci_register_driver(t4_driver); That allows you to get the card working on 2.6.13 in almost exactly the same setup as yours. One weird thing though. Do no use insmod ./wct4xxp.ko from zaptel directory as it will not work. Do a proper make install and then modprobe. This is just part of the fixes you might need to do. If you encounter a problem after span reconfiguration (ztcfg) let me know. If you get stuck.. let me know. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Sunday, September 11, 2005 8:14 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] TE406p no interrupts I'm using FC3. uname -a - Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2 15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! for Good Watch the Hurricane Katrina Shelter From The Storm concert http://advision.webevents.yahoo.com/shelter ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] TE406p no interrupts
My motherboard is TYAN Tiger K8W. I was happy with this board and previous te405p, except some echo issue. I will try on 2.6.13 on 955X chipset. But I am not an expert on linux. So I want to know easier way. Any successful installation on FC4? If someonw know any success story of te406p, please share with me. Regard --- Boris Bakchiev [EMAIL PROTECTED] wrote: Well. That means pci_register_driver probably not ding what it supposed to do. In newer kernels pci_module_init should be replaced with pci_register_driver as pci_module_init doesn't it what it supposed to. How brave are you at getting a new kernel on your system? I'm currently running on 2.6.13 on 955X chipset and it works really well. At first I had all sorts of problems with interrupts but with couple of patches to wct4xxp all working just fine with close to 3-5K of calls per day. What is the model of the motherboard you have? See if you can force a particular IRQ on a slot where your TE406P is. Some motherboards do allow this, so you can assign IRQ bellow 15 to the card. That could help as well. For now, revert the changes back. If you can, try new kernel (in parallel) with the pci_register_driver. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Monday, 12 September 2005 11:28 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] TE406p no interrupts I modified wct4xxp.c and installed it. This is the message for 'modprobe wct4xxp' -- FATAL: Error inserting wct4xxp (/lib/modules/2.6.9-1.667smp/extra/wct4xxp.ko): No such device FATAL: Error running install command for wct4xxp astpbx kernel: Oops: [1] SMP astpbx kernel: CR2: a0362081 Regards, Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! for Good Watch the Hurricane Katrina Shelter From The Storm concert http://advision.webevents.yahoo.com/shelter ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] PRI echo
Hi, My configuration is pri*(te405p)---iaxclient. My * version is 1.0.7 running on tyan dual opteron board. I have several problems. 1) inbound echo For outbound call(iaxclient--pri), there is almost no echo. But for inbound(pri--iaxclient), I can hear distinct echo. Can Sangoma a104 or digium te406p help this problem? 2)Today i received te406p. I know T1/E1 jumper. But how can i change the configuration of te406p for echo cancel mode selection? 3) asterisk crash (gdb) bt #0 0x002a973f15dd in q921_transmit_iframe (pri=0x2a9cd0ecf0, buf=0x40ffeac0, len=9, cr=1) at q921.c:384 #1 0x002a973f701c in q931_xmit (pri=0x2a9cd0ecf0, h=0x40ffeac0, len=9, cr=1) at q931.c:1848 #2 0x002a973f720f in send_message (pri=0x2a9cd0ecf0, c=0x2a9cd12810, msgtype=77, ies=0x2a97500570) at q931.c:1888 #3 0x002a973f7b31 in q931_release (pri=0x2a9cd0ecf0, c=0x2a9cd12810, cause=16) at q931.c:2141 #4 0x002a973f78eb in pri_disconnect_timeout (data=0x2a9cd12810) at q931.c:2092 #5 0x002a973f309b in __pri_schedule_run (pri=0x2a95aa1da0, tv=0x40ffefa0) at prisched.c:97 #6 0x002a973f30f8 in pri_schedule_run (pri=0x2a95aa1da0) at prisched.c:109 #7 0x002a9729e77a in pri_dchannel (vpri=0x2a9cd0ecf0) at chan_zap.c:7415 #8 0x00307f305f81 in start_thread () from /lib64/tls/libpthread.so.0 #9 0x00307e6c3af3 in thread_start () from /lib64/tls/libc.so.6 #10 0x in ?? () 4)chan_iax2 Some times asterisk log file is filled with strange message as follow. Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! And some times it's overloading my sata HDD and disable ssh connection. iax.conf --- [general] port=5036 disallow=all tos=0x04 qualify=no [agent] type=friend username=agent secret=agent context=agent host=dynamic notransfer=yes callerid=20005000 zapata.conf --- [channels] context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national nationalprefix= signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes callprogressdetect=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=2.0 txgain=-4.0 group=1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 -- Thanks, have a great holiday! Regards, Jason __ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] I should never be called!
Hi, My configuration is pri - * - iaxclient based phone. Almost every day asterisk log file is filled with strange lines as follow. - Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! -- My * version is 1.0.7 running on tyan dual opteron board and 2 E1 sangoma card. iax.conf --- [general] port=5036 disallow=all tos=0x04 qualify=no [agent] type=friend username=agent secret=agent context=agent host=dynamic notransfer=yes callerid=20005000 zapata.conf --- [channels] context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national nationalprefix= signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes callprogressdetect=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=2.0 txgain=-4.0 group=1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 -- Anyone please help? Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] NIC irq load balancing
Hi All, I'm developing an outbound call center with 20 agents. My configuration is like this. PRI * NetGear Switch 20 iaxSoftPhone I'm experincing bad voice quality and long delay. I'm thinking about several possibilities. 1. NIC load - All NIC irqs process by CPU0. I tried irabalance, but no effect. #cat /proc/interrupts CPU0 CPU1 0: 158232 385091IO-APIC-edge timer 1: 3 0IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 3: 56271IO-APIC-edge serial 4: 10 0IO-APIC-edge serial 8: 1 0IO-APIC-edge rtc 12:185 0IO-APIC-edge PS/2 Mouse 14: 20312 2667IO-APIC-edge libata 15: 0 0 XT-PIC libata 17: 573544 98848 IO-APIC-level Intel ICH5 18: 947385 0 IO-APIC-level eth0 21: 7908154593392 IO-APIC-level t1xxp NMI: 0 0 LOC: 543220 543219 ERR: 0 MIS: 0 2. NetGear Switch - I'm using FS-526T Switch, which has 24 10/100 ports and 2 Gb sorts. I want to know if this kind of general purpose switch is not suitable for voip. If so, could you recommand one? 3. Server - My server is based on ASUS md, 2 Xeon 2.8G, 1GB ram, 1 sata drive. OS is Redhat9.0 CPU's idle status is 70~100. Regards, Jason __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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
