RE: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute

2006-10-05 Thread Andrew Shelton 








What codec are you using?



How many phone? What load is the server
under?















From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 13:22
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Extremely choppy sound on some of our POTSnetwork calls; goes away with mute





1)Can anyone tell me how to do this on a Polycom 501?

2)Can you explain why you think this any why it ony happens on some calls?

Thanks

Andres
[EMAIL PROTECTED] wrote:




 For about 20% of the calls to the outside world, the voice on the 
 other end of an outside line is incredibly choppy. Enough to where 
 we have to hang up and call on a cell phone. It is always the same 
 numbers that are choppy. The funny thing is, if I press mute while 
 talking on a choppy call, the choppiness goes away completely.

 

Maybe you have silence suppression enabled on your phones. Try to 
disable it and see if it helps.



 



-- 
Andres
Technical Support
http://www.telesip.net

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[asterisk-users] GXP - 2000 BLF

2006-10-05 Thread Andrew Shelton 








Hello,



I have been trying to get my Grandstream busy line filter to
work for ages..



All the lights flash as they are supposed to.



If one Grandstream 7000 calls another Grandstream 7003 I can
use Grandstream 7002 to pick the call up pressing the BLF button and all works
fine.



However if I call Grandstream 7000 with a mobile phone and
try to pickup the call with Grandstream 7002 all I get is a 603 error on
Grandstream 7002.



I am using firmware 1.1.12 for the Grandstream and 1.2.12.1
version of asterisk





This is the error I get from my log..



if some one could please help

Oct 5 12:12:51 DEBUG[7723] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing NoOp(SIP/7003-b721be28, **7002) in new stackOct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing Pickup(SIP/7003-b721be28, 7002) in new stackOct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No originating channel found.Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup possible...Oct 5 12:12:51 VERBOSE[8828] logger.c: == Spawn extension (inbound-from-stem, **7002, 2) exited non-zero on 'SIP/7003-b721be28'Oct 5 12:12:51 DEBUG[7716] channel.c: Avoiding initial deadlock for 'SIP/7003-b721be28'



SIP[7000]type=friendcontext=inbound-from-stemSubscribecontext=BLFsecret=*host=dynamiccanreinvite=nocallgroup=2pickupgroup=2[EMAIL PROTECTED]username=7000dtmfmode=rfc2833callerid=STEM 17524543545qualify=yes





EXTENSIONS



[default]

include = stem

include = to-siemens

include = BLF

include = BLF_group_pickup





[stem]

;exten STEM GROUP = 01752 692205

exten = 123454,1,Ringing

exten = 123454,n,Wait(1)

exten = 123454,n,Answer()

exten = 123454,n,NoOp(${CALLERID(all)})

exten = 123454,n,SetCIDName(Outside Caller)

exten = 123454,n,Set(CALLERID(number)=9${CALLERIDNUM})

exten = 123454,n,NoOp(${CALLERID(all)})

exten = 123454,n,Macro(stdexten2,7003,${STEMGROUP},20)



;exten 7000 = 01752 692204

exten = 123455,1,Ringing

exten = 123455,n,Wait(1)

exten = 123455,n,Answer()

exten = 123455,n,NoOp(${CALLERID(all)})

exten = 123455,n,SetCIDName(Outside Caller)

exten = 123455,n,Set(CALLERID(number)=9${CALLERIDNUM})

exten = 123455,n,NoOp(${CALLERID(all)})

exten = 123455,n,Macro(stdexten2,7000,${stem},20)



;exten 7001 = 01752 692283

exten = 123456,1,Ringing

exten = 123456,n,Wait(1)

exten = 123456,n,Answer()

exten = 123456,n,NoOp(${CALLERID(all)})

exten = 123456,n,SetCIDName(Outside Caller)

exten = 123456,n,Set(CALLERID(number)=9${CALLERIDNUM})

exten = 123456,n,NoOp(${CALLERID(all)})

exten = 123456,n,Macro(stdexten2,7001,${stem1},20)





[internal]

;Internal Extensions

exten = _7XXX,1,Ringing

exten = _7XXX,n,Wait(1)

exten = _7XXX,n,Answer()

exten = _7XXX,n,Set(FOO1=${CHANNEL:4})

exten = _7XXX,n,Set(FOO2=${CUT(FOO1,-,1)})

exten = _7XXX,n,Set(CALLERID(number)=${FOO2})

exten = _7XXX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN})





[inbound-from-pstn] ; inbound calls to this context from
outside lines

include = default





[inbound-from-sip]

include = default



[inbound-from-local]

;from sip default context used.. requires hints

include = voicemail

include = provider

include = outbound

;include = stem ;for hints





[inbound-from-stem]

include = BLF

include = internal

include = DefExt

include = voicemail

include = outbound

include = BLF_group_pickup

include = feature-cfu

include = feature-cfna

include = feature-cfb



[inbound-from-logicall]

include = internal

include = DefExt

include = voicemail

include = outbound

include = BLF_group_pickup

include = feature-cfu

include = feature-cfna

include = feature-cfb



;Test section for BLF on Grandstreams for Stem

[BLF_group_pickup]

include =inbound-from-stem

exten = _**.,1,NoOp(${EXTEN})

exten = _**.,2,Pickup(${EXTEN:2})

exten = _**.,3,Hangup



[BLF]

include =inbound-from-stem

exten =7000,hint,SIP/7000

exten =7000,1,Dial(SIP/7000,20,r)

exten =7001,hint,SIP/7001

exten =7001,1,Dial(SIP/7001,20,r)

exten =7002,hint,SIP/7002

exten =7002,1,Dial(SIP/7002,20,r)

exten =7003,hint,SIP/7003

exten =7003,1,Dial(SIP/7003,20,r)

exten =7004,hint,SIP/7004

exten =7004,1,Dial(SIP/7004,20,r)

exten =7005,hint,SIP/7005

exten =7005,1,Dial(SIP/7005,20,r)

exten =7006,hint,SIP/7006

exten =7006,1,Dial(SIP/7006,20,r)






































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RE: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute

2006-10-05 Thread Andrew Shelton 








Well I am using GSM as my main codec which
seems to be very nice

I would also suggest you looking at the
load of you CPU I know that asterisk is very processor hungry



You can also change some settings in the
zapta and zaptel config.. to reduce echo and interference on the line..











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 14:38
To:
asterisk-users@lists.digium.com
Subject: RE: [asterisk-users]
Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute





Below is the text of my
original post. I am not sure what Codec we are using. The Codec
Preferences phone setting shows, in order of preference, G.711u, G.711A,
G.729AB



We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core
4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with
2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium
TDM400P card which is connected to 4 POTS lines. The server is also
connected to a 100MB switched LAN where we have about 20 Polycom 501 phones
with the latest firmware updates. Nothing else runs on the server except an ftp
daemon which is never used except when a phone reboots.

For about 20% of the calls to the outside world, the voice on the other end of
an outside line is incredibly choppy. Enough to where we have to
hang up and call on a cell phone. It is always the same numbers that are
choppy. The funny thing is, if I press mute while talking on a choppy
call, the choppiness goes away completely.





I have tried: turning off ACPI, turning off APCI, moving the card to
another PCI slot, changing the RX/TX gains. There are no shared IRQs. I have
tested the lines by unplugging them from the asterisk server and plugging them
directly into an analogue phone. Using cat /proc/interrupts; sleep 10 ;
cat /proc/interrupts I see that there are about 1,000 interrupts per
seconds between the card and the CPU.





I do not think it is a network congestion problem as intra-office
communications as well as voicemail retrieval are always perfect. The Voip does
not go over any routers, just a max of 2 switches with a 1GB trunk. This
happens even off-hours when the network isnt being used at all.





There are never more than 2 people on the phone at the same time and it
is definitely not an over-utilized processor.





I have trying to figure
this out for 2 months on and off with no success any help is appreciated.





Thanks

Andrew Shelton
[EMAIL PROTECTED] wrote:



What
codec are you using?











How many phone? What load is the server
under?































From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 13:22
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Extremely choppy sound on some of our POTSnetwork calls; goes away with mute













1)Can anyone tell me how to do this on a Polycom 501?

2)Can you explain why you think this any why it ony happens on some calls?

Thanks

Andres
[EMAIL PROTECTED] wrote:








 For about 20% of the calls to the outside world, the voice on the 
 other end of an outside line is incredibly choppy. Enough to where 
 we have to hang up and call on a cell phone. It is always the same 
 numbers that are choppy. The funny thing is, if I press mute while 
 talking on a choppy call, the choppiness goes away completely.

 

Maybe you have silence suppression enabled on your phones. Try to 
disable it and see if it helps.



 



-- 
Andres
Technical Support
http://www.telesip.net

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Yahoo! Messenger with Voice. Make
PC-to-Phone Calls to the US
(and 30+ countries) for 2¢/min or less.



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RE: [asterisk-users] two asterisk and one NBX system

2006-10-05 Thread Andrew Shelton 
I would research the switch statement and DUNDI

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jose diaz
Sent: 05 October 2006 14:51
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] two asterisk and one NBX system

We have three servers: Two asterisk and one NBX 3COM.
The connection between Asterisk1 and Asterisk2 is with IAX2.
The connection between  Asterisk2 and NBX is with a Digium analog 
TDM400P (2FXO and 2 FXS)

The dial plan Asterisk1: 3XXX
The dial plan Asterisk2: 2XXX
The dial plan NBX: 1XXX

The system work well, but the call from Asterisk1 to NBX fail. I'm using

the IAX2 protocol to call from asterisk1 to asterisk2, i need to 
trasnfer the call to the NBX. How i can to make that?

Regards,

Jose Diaz

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