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. ___ -- 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
Re: [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: [EMAIL PROTECTED] To: asterisk-users@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___ -- 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
Re: [asterisk-users] sip conversations overlapping!!!!
Remove pstn lines from sipura and call sipura to sipura ... any problems ? Still with pstn lines removed call sipura1 sipura2 and after sipura 3sipura1 do you still hear any voices? if not it's you cable to pstn. Give us feedback ___ -- 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
Re: [asterisk-users] sip conversations overlapping!!!!
RoLaNd RoLaNd wrote: 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) Lines plugged into the fxS ports? I hope you have them in the LINE ports (fxO) Are there any telephones plugged directly into the Sipuras? Into the PHONE ports (FXS) 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.. I would double check the wiring of the 8 line cable. 4 POTS lines = 4 pairs of tip ring. Are there some of the tipring pairs mixed up? eg tip from line 1 mixed with ring from line 2, etc. This is the most likely scenario since I can't imagine Asterisk bridging the calls without being asked to. Otherwise, are we still missing something in the topology here? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] 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/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(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
[EMAIL PROTECTED]
canreinvite=yes
[301]
type=friend
host=dynamic
secret=1234
context=sipura-line
[EMAIL PROTECTED]
[304]
type=friend
host=dynamic
secret=1234
context=sipura-line2
[EMAIL PROTECTED]
[305]
type=friend
host=dynamic
secret=1234
context=incoming-samer
[EMAIL PROTECTED]
[306]
type=friend
host=dynamic
secret=1234
context=incoming-gilberte
[EMAIL PROTECTED]
[333]
type=friend
host=dynamic
secret=1234
context=incoming-conference
[EMAIL PROTECTED]
[334]
type=friend
host=dynamic
secret=1234
context=incoming-samer
[EMAIL PROTECTED]
[335]
type=friend
host=dynamic
secret=1234
context=incoming-gilberte
[EMAIL PROTECTED]
[336]
type=friend
host=dynamic
secret=1234
context=incoming-line4
[EMAIL PROTECTED]
[307]
type=friend
host=dynamic
secret=1234
context=incoming-conference
[EMAIL PROTECTED]
[308]
type=friend
host=dynamic
secret=1234
context=incoming-line4
[EMAIL PROTECTED]
[310]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[320]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[330]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[340]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[350]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[107]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[150]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[100]
type=friend
host=dynamic
Re: [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 ___ -- 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
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! Date: Thu, 28 Aug 2008 14:10:53 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [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 ___ -- 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 _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- 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
