thr3ads.net - asterisk users - [asterisk-users] sip conversations overlapping!!!! [Aug 2008]

If this information is useful, please help other people find it:
Share via:

RoLaNd RoLaNd

2008-Aug-28 17:12 UTC

[asterisk-users] sip conversations overlapping!!!!

Hi all,



i'm facing this weird prob...my topology is as such:



<softphone< --- <asterisk> ---- <sipura 3102>

                                        
----<sipura 3102>

                                       
-----<sipura 3102>

                                       
-----<sipura 3102>



when am on a call, sometimes when some1 else tries to call out.. i hear the
actual tones which ends up preventing the other end from talking to me.. 

moroever, when some1 calls me through one sipura, while im talking on another...
i can hear the attendant welcoming message, then i hear the voice of whoever
have picked tht line up..! and if i ask that person to hang up... my line breaks
as well..!

can any1 help me with this issue! 

below is my config:



extensions.conf



[incoming-conference]

exten => 333,1,Answer() ; Answer inbound calls

exten => 333,2,Playback(silence/1)

exten => 333,3,Background(joyce) ; input an extension

exten => 333,4,WaitExten(8)

exten => 333,5,Dial(SIP/310,15)

exten => 333,4,Wait(8)

include => spa

exten => 333,n,Hangup()



[incoming-samer]

exten => 334,1,Answer() ; Answer inbound calls

exten => 334,2,Playback(silence/1)

exten => 334,3,Background(joyce) ; input an extension

exten => 334,4,WaitExten(8)

exten => 334,5,Dial(SIP/330,15)

exten => 334,4,Wait(8)

include => spa

exten => 334,n,Hangup()



[incoming-gilberte]

exten => 335,1,Answer() ; Answer inbound calls

exten => 335,2,Playback(silence/1)

exten => 335,3,Background(joyce) ; input an extension

exten => 335,4,WaitExten(8)

exten => 335,5,Dial(SIP/350,15)

exten => 335,4,Wait(8)

include => spa

exten => 335,n,Hangup()



[incoming-line4]

exten => 336,1,Answer() ; Answer inbound calls

exten => 336,2,Playback(silence/1)

exten => 336,3,Background(joyce) ; input an extension

exten => 336,4,WaitExten(8)

exten => 336,5,Dial(SIP/340,15)

exten => 336,4,Wait(8)

include => spa

exten => 336,n,Hangup()





[sipura-line]

exten => 301,1,Answer() ; Answer inbound calls

exten => 301,2,Playback(silence/1)

exten => 301,3,Background(simzy1) ; input an extension

exten => 301,4,WaitExten(8)

exten => 301,5,Dial(SIP/100,15) ; goes to operator

exten => 301,4,Wait(8)

include => spa



[spa]

exten =>_301,1,GoTo(sipura-line,${EXTEN},1)

exten =>_333,1,GoTo(incoming-conference,${EXTEN},1)

exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times

exten => _1XX,2,VoiceMail(${EXTEN}@default) ; direct 2 voicemail box if line
is busy or unavailable

exten => _1XX,3,HangUp()

exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times

exten => _2XX,2,VoiceMail(${EXTEN}@default) ; directs to voicemail box if
line is busy or unavailable

exten => _2XX,3,HangUp()

exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times

exten => _3XX,2,VoiceMail(${EXTEN}@default) ; directs 2 voicemail box if
line is busy or unavailable

exten => _3XX,3,HangUp()

exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line

;exten =>_01,2,Set(TIMEOUT(absolute)=5)

exten =>_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line

exten =>_03,1,Dial(SIP/$(EXTEN)@305) ; samer 

exten =>_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte 

exten =>_05,1,Dial(SIP/$(EXTEN)@307) ; conference 

exten =>_06,1,Dial(SIP/$(EXTEN)@308) ; line 4 

exten => 303,1,VoicemailMain ; voicemail box to be redirected to







sip.conf:



[300]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=300 at default

canreinvite=yes



[301]

type=friend

host=dynamic

secret=1234

context=sipura-line

mailbox=301 at default



[304]

type=friend

host=dynamic

secret=1234

context=sipura-line2

mailbox=304 at default



[305]

type=friend

host=dynamic

secret=1234

context=incoming-samer

mailbox=305 at default



[306]

type=friend

host=dynamic

secret=1234

context=incoming-gilberte

mailbox=306 at default





[333]

type=friend

host=dynamic

secret=1234

context=incoming-conference

mailbox=333 at default



[334]

type=friend

host=dynamic

secret=1234

context=incoming-samer

mailbox=334 at default



[335]

type=friend

host=dynamic

secret=1234

context=incoming-gilberte

mailbox=335 at default





[336]

type=friend

host=dynamic

secret=1234

context=incoming-line4

mailbox=336 at default



[307]

type=friend

host=dynamic

secret=1234

context=incoming-conference

mailbox=307 at default



[308]

type=friend

host=dynamic

secret=1234

context=incoming-line4

mailbox=308 at default



[310]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=310 at default



[320]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=320 at default



[330]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=330 at default



[340]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=340 at default



[350]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=350 at default





[107]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=107 at default



[150]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=150 at default





[100]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=100 at default



[172]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=172 at default



[101]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=101 at default





[102]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=102 at default





[103]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=103 at default





[104]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=104 at default





[105]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=105 at default





[106]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=106 at default







[108]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=108 at default



[109]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=109 at default





[110]

type=friend

host=dynamic

secret=1234

context=spa

mailbox=110 at default




_________________________________________________________________
Connect to the next generation of MSN Messenger?
http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080828/9edd222e/attachment.htm

Drew Gibson

2008-Aug-28 18:10 UTC

head link

[asterisk-users] sip conversations overlapping!!!!

RoLaNd RoLaNd wrote:>
> Hi all,
>
> i'm facing this weird prob...my topology is as such:
>
> <softphone< --- <asterisk> ---- <sipura 3102>
>                                          ----<sipura 3102>
>                                         -----<sipura 3102>
>                                         -----<sipura 3102>
>
> when am on a call, sometimes when some1 else tries to call out.. i 
> hear the actual tones which ends up preventing the other end from 
> talking to me..
> moroever, when some1 calls me through one sipura, while im talking on 
> another... i can hear the attendant welcoming message, then i hear the 
> voice of whoever have picked tht line up..! and if i ask that person 
> to hang up... my line breaks as well..!
> can any1 help me with this issue!
> below is my config:
>
How are the analogue phones wired? One phone plugged directly to one 
3102 FXS port? or is there common wiring ?
Are all the FXO ports connected to telco lines?


regards,

Drew

NOTE: Holding the <SHIFT> key down whilst typing the first person, 
singular, pronoun will produce stunningly readable results. Either 
<SHIFT> key will do, you can even use the <CAPS LOCK> key if both of
those are broken/can't locate them. You can also use this procedure for 
the first letter of each sentence, it makes everything much easier to read.

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

RoLaNd RoLaNd

2008-Aug-29 09:26 UTC

head link

[asterisk-users] sip conversations overlapping!!!!

i appologize for not making myself clear..


i have my asterisk box, connexted to 4 sipura3102..
 these sipuras has 4 PSTN lines connected to them through one cable, which has 8
lines inside of it (2 connected to an RJ11 and plugged into its respecitve fxs
port in the sipura)
on the other side, i have 20 softphones.. these softphones has asterisk as their
gateway.. where they could call eachother! or call/recieve calls through any of
the sipuras...


my prob is as such:

when i call from softphone#1 to sipura #1, sound is pretty good and everything
is working perfectly.. though if asterisk recieves a call from another sipura..
lets say its sipura #2, then! i could hear the attendnat answering the incoming
phone in my current conversation, and i could hear some1 picking up and
answerinfg the call..!
if i ask them to hang up! my line breaks as well..



Date: Fri, 29 Aug 2008 10:40:57 +0200
From: megahohol at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] sip conversations overlapping!!!!
>Every one PSTN line connected to the FXS port of sipura..
>Though these 4 lines comes in one cable if that has to do with anything!
Not clear for me, develop some more you topology.


_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080829/54d89f35/attachment.htm

asterisk users - Aug 2008 - sip conversations overlapping!!!!

[asterisk-users] sip conversations overlapping!!!!

[asterisk-users] sip conversations overlapping!!!!

[asterisk-users] sip conversations overlapping!!!!