Re: [asterisk-users] asterisk-gui-2.1.0-rc1
Hi Yes , I installed it from the source. As you suggested, I re installed it from SVN. When I try to execute ./configure , got the following message , configure: error: *** JSON support not found (this typically means the libjansson development package is missing) After that I tried to install certain packages by, yum groupinstall 'Development Tools' and yum install php-devel php-pear php-common yum install json.so but , no luck . still i am getting above message. Thanks in advance Luke From: aristidis tsitras To: asterisk-users@lists.digium.com Sent: Saturday, 25 May 2013, 2:31 Subject: Re: [asterisk-users] asterisk-gui-2.1.0-rc1 Hi, how did you installed it? if it is svn, thry to install it again. if it is through source then delete it and try through svn Hi > >I have installed asterisk-gui-2.1.0-rc1 . After I logged in to the GUI , it >was continuously refreshing the web browser and trying to load the >configurations. > > >Can I know where is gone wrong ? > > >Thanks in advance >Luke > > >-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Solaris
Bump On 5/23/13, Nick Khamis wrote: > Hello Everyone, > > I have bumped into the thralling penguin page on linux vs solaris for > asterisk. Does the benchmark still hold with the newer versions of > kernels? Curious to know of your thoughts. Also, they mentioned > running it on Sun Fire x2100, but no benchmarks were given for that. > > Can increased performance be accomplished simply by changing to > Solaris or OpenSolaris? > > > Kind Regards, > > Nick. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-gui-2.1.0-rc1
Try to use firefox instead of IE. Besides, you may check if there is any problem in the extensions.conf. My recent experiment of installing gui into asterisk 11.x is that there is problem in some of the macro script within extensions.conf. I delete the sample macro scripts in extensions.conf and use the attached for my asterisk. On Sat, May 25, 2013 at 2:31 AM, aristidis tsitras wrote: > Hi, how did you installed it? > if it is svn, thry to install it again. > if it is through source then delete it and try through svn > > > Hi > > > I have installed asterisk-gui-2.1.0-rc1 . After I logged in to the GUI , > it was continuously refreshing the web browser and trying to load the > configurations. > > Can I know where is gone wrong ? > > Thanks in advance > Luke > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > extensions_macro.conf Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 dtmf not recognised
Tried info, rfc2833, inband and finally kept as auto. On 25 May 2013 02:20, "Doug Lytle" wrote: > >> dtmfmode=auto > > dtmfmode=info > > or > > dtmfmode=rfc2833 > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 dtmf not recognised
>> dtmfmode=auto dtmfmode=info or dtmfmode=rfc2833 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 dtmf not recognised
Hi
I have a dialplan as per the following,
extensions.conf
[avgtest]
exten = 100,n,Playback(avgtest/message1)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,,5,3) ; Attempts for 5 times with 3 seconds of
timeout
exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
exten = 100,n,Hangup() ; Didn't go to pin-accepted, so play badPIN and
hangup
exten=pinaccepted,1,Playback(avgtest/message2) ; correct pin, play
sipconf
[1001]
uername=1001
secret=1001
context=avgtest
disallow=all
allow=ulaw
allow=alaw
dtmfmode=auto
type=friend
host=dynamic
canreinvite=yes
relaxdtmf=yes
This looks very simple but dtmf is not recognised.
Am using asterisk 11.
Any suggestions is much appreciated.
Regards
--
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Re: [asterisk-users] asterisk-gui-2.1.0-rc1
Hi, how did you installed it? if it is svn, thry to install it again. if it is through source then delete it and try through svn Hi I have installed asterisk-gui-2.1.0-rc1 . After I logged in to the GUI , it was continuously refreshing the web browser and trying to load the configurations. Can I know where is gone wrong ? Thanks in advance Luke -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-gui-2.1.0-rc1
Hi I have installed asterisk-gui-2.1.0-rc1 . After I logged in to the GUI , it was continuously refreshing the web browser and trying to load the configurations. Can I know where is gone wrong ? Thanks in advance Luke -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pri-Debug-Log / Is Early Media supported by provider?
Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the following: (I replaced originating caller id by 123456) PRI Span: 1 < Protocol Discriminator: Q.931 (8) len=48 PRI Span: 1 < TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent from originator) PRI Span: 1 < Message Type: SETUP (5) PRI Span: 1 < [a1] PRI Span: 1 < Sending Complete (len= 1) PRI Span: 1 < [04 03 80 90 a3] PRI Span: 1 < Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1 < Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1DL-DATA request PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=10 PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 > Message Type: CALL PROCEEDING (2) PRI Span: 1 TEI=0 Transmitting N(S)=70, window is open V(A)=70 K=7 PRI Span: 1 PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=10 PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 > Message Type: CALL PROCEEDING (2) PRI Span: 1 > [18 03 a9 83 8e] PRI Span: 1 > Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 PRI Span: 1 > ChanSel: As indicated in following octets PRI Span: 1 > Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1 > Ext: 1 Channel: 14 Type: CPE] -- Accepting call from '123456' to '1' on channel 0/14, span 1 -- Executing [1@port1:1] NoOp("DAHDI/i1/123456-245", "") in new stack -- Executing [1@port1:2] Playback("DAHDI/i1/123456-245", "demo-thanks,noanswer") in new stack -- Playing 'demo-thanks.gsm' (language 'de_female') -- Executing [1@port1:3] Hangup("DAHDI/i1/123456-245", "") in new stack == Spawn extension (port1, 1, 3) exited non-zero on 'DAHDI/i1/123456-245' PRI Span: 1 q931.c:6837 q931_hangup: Hangup other cref:14783 PRI Span: 1 q931.c:6594 __q931_hangup: ourstate Incoming Call Proceeding, peerstate Outgoing Call Proceeding, hold-state Idle PRI Span: 1 q931.c:5783 q931_disconnect: Call 14783 enters state 11 (Disconnect Request). Hold state: Idle PRI Span: 1 PRI Span: 1 > DL-DATA request PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 > Message Type: DISCONNECT (69) PRI Span: 1 TEI=0 Transmitting N(S)=71, window is open V(A)=71 K=7 PRI Span: 1 PRI Span: 1 > Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to originator) PRI Span: 1 > Message Type: DISCONNECT (69) PRI Span: 1 > [08 02 81 90] PRI Span: 1 > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network servin
Re: [asterisk-users] Registration timed out - for created users
you don't need register => string here, it only need you want asterisk register to another sip proxy as client. just remove that line and you should fine. for X-lite or any other sip phone the user "AlphaUser" is sufficient. On Fri, May 24, 2013 at 12:32 PM, luke devon wrote: > > Hi all , > > I have managed to install and configure the > > 1. asterisk-1.8-current > 2. dahdi-linux-complete-current > > > I did not faced any issues during the installation. After that I installed > X-Lite soft phone in two different PCs and tested the setup. every thing > was success. I was able make calls from each extensions. > > > But when I observe the log files , i could see some messages .. > > chan_sip.c:-- Registration for 'alphaUser@192.168.1.12' timed out, > trying again (Attempt #2) > > Something is not right. I have double check the configurations. But I > could not find where I have done the mistake. > > following is my configurations, > > sip.conf > --- > register => alpahaUser:1234@192.168.1.10 > > [alphaUser] > type=friend > username=alphaUser > secret=1234 > context=tutorial > host=dynamic > canreinvite=no > dtfmode=rfc2833 > disallow=all > allow=ulaw > subscribecontext=tutorial > mailbox=alphaUser@internal > > > extensions.conf > > [tutorial] > exten => ,1,Dial(SIP/alphaUser) > > > Please help me to identify and resolve the issue . > > Thanks in Advance > Luke. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registration timed out - for created users
Hi all , I have managed to install and configure the 1. asterisk-1.8-current 2. dahdi-linux-complete-current I did not faced any issues during the installation. After that I installed X-Lite soft phone in two different PCs and tested the setup. every thing was success. I was able make calls from each extensions. But when I observe the log files , i could see some messages .. chan_sip.c: -- Registration for 'alphaUser@192.168.1.12' timed out, trying again (Attempt #2) Something is not right. I have double check the configurations. But I could not find where I have done the mistake. following is my configurations, sip.conf --- register => alpahaUser:1234@192.168.1.10 [alphaUser] type=friend username=alphaUser secret=1234 context=tutorial host=dynamic canreinvite=no dtfmode=rfc2833 disallow=all allow=ulaw subscribecontext=tutorial mailbox=alphaUser@internal extensions.conf [tutorial] exten => ,1,Dial(SIP/alphaUser) Please help me to identify and resolve the issue . Thanks in Advance Luke.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error 488 Not Acceptable Here
Hi, Am Donnerstag, den 23.05.2013, 20:48 +0200 schrieb Maximilian Grobecker: > Am 22.05.2013 16:39, schrieb Andrew Colin: > > Hi guys, > > > > Any idea why I am getting this error when someone tries to send me a T38 > > Fax? > Hi, > > Maybe you have not allowed T.38 as acceptable codec ;-) > You can try with "allow=all" in your sip.conf. No, T.38 is not a codec and so allow=all will not help. To use T.38 You have to enable T.38 with "t38pt_udptl = yes" in sip.conf. The reason, why You get a "488 Not Acceptable Here488 Not Acceptable Here", is only detectable with a SIP Trace. There are many reasons e.g. - Your asterisk version does not support T.38 - T.38 is not enabled (see above) - T.38 is enabled, but not at the relevant peers (in most versions of asterisk there is only support for T.38 passthrough, so both peers have to support T.38) - There are problems in the transmission and some peers wants to switch back to audio level and the other or asterisk is not willing to support this. The last reason may occur, if You have NAT and do not correctly forward the data ports of T.38 (UDPTL Ports). Best way is to get a SIP Trace to analyse. If You provide one, You should also tell, which version of asterisk. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
