Re: [asterisk-users] Play audio file for both Caller and Callee in a call
Hi all,
I made it easier, AMI was not required, it can be solved directly in the
dialplan:
same = n,Dial(SIP/${TRUNK}/${ARG2}${NUM},60,M(inject^${CALLERNUMBER}))
[macro-inject]
same = s,1,Originate(Local/trunk@injectWarning,app,Playback,Message-Callee)
same = n,Originate(Local/${ARG1}@injectWarning,app,Playback,Message-Caller)
[injectWarning]
exten = _+34[69],1,Answer()
same = n, ChanSpy(SIP/${EXTEN},qw)
same = n, Hangup()
exten =trunk,1,Answer()
same = n, ChanSpy(SIP/${TRUNK},qw)
same = n, Hangup()
Thank you all!
Date: Thu, 15 Dec 2011 23:56:15 -0800
From: Jim Dickenson dicken...@cfmc.com
Subject: Re: [asterisk-users] Play audio file for both Caller and
Callee in a call
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 92e76a79-3929-4978-82f2-ee8c1db50...@cfmc.com
Content-Type: text/plain; charset=windows-1252
Use an AMI packet like this:
Action: Originate
Channel: Local/do_playback@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280
ActionID: PlayBack
Async: true
With dialplan like this:
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}) exten =
do_playback,n,Wait(0.3) exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS}) exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}) exten =
do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}) exten =
do_chanspy,n,Hangup()
You need to issue an AMI packet for each leg of the call. Each leg will hear
the same audio feed offset by however long it takes the packets to be
processed. In general this is a few milliseconds and should not be a big deal.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
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[asterisk-users] AMI and Dialplan
Hello all, This may sound an odd question but if you initiate a call using AMI does it adhere to what has been defined in the dial plan or do we have to write the logic into the AMI call ? -- Thanks, Phil -- _ -- 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] AMI and Dialplan
Please ignore as this was a user error! -- Thanks, Phil - Original Message - Hello all, This may sound an odd question but if you initiate a call using AMI does it adhere to what has been defined in the dial plan or do we have to write the logic into the AMI call ? -- Thanks, Phil -- _ -- 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] No rtpmap codec info in 200 OK
Ok. Asterisk sends the rtpmap info for the codec. Is it possible to remove this from the 200 OK sent by Asterisk? Possible direction I should look. -- _ -- 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] Sending Fax Dialplan with Retry Attempt
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when sending Fax
are very Low especially when we send the Fax just once. Now I’m trying to
modify the dialplan so it will keep trying to send the fax for maximum 5 times
at once and it only retry if the Sending Status has Error in it.
Here is the dialplan :
[fax-tx]
;Fax For Asterisk - Digium
exten = s,1,NoOp( SENDING FAX )
exten = s,n,Wait(6)
;zoel : Insert to MySQL
exten = s,n,MySQL(Connect connid localhost asterisk)
exten = s,n,GotoIf($[${MYSQL_STATUS} = 0]?:mysql_error,1)
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO fax_activity SET
date=now(),faxfile=${FAXFILE},faxtxnumber=${FAXTXNUMBER},faxresult=Sent to
Spooler,faxpages=${FAXPAGES},faxstatus=${FAXERROR},faxline=${CHANNEL})
exten = s,n,MYSQL(Disconnect ${connid})
;zoel : End Insert to MySQL
; Set FAXOPTs
exten = s,n,NoOp( SETTING FAXOPT )
exten = s,n,Set(FAXOPT(ecm)=yes)
exten = s,n,Set(FAXOPT(headerinfo)=TEST FAX)
exten = s,n,Set(FAXOPT(localstationid)=1234567)
exten = s,n,Set(FAXOPT(maxrate)=14400)
exten = s,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 1 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 2
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here … L
exten = s,n,Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 2 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 3
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt3)
exten = s,n(RetryAttempt3),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 3 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 4
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt4)
exten = s,n(RetryAttempt4),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 4 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 5
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt5)
exten = s,n(RetryAttempt5),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 5 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
; Hangup! Print FAXOPTs
exten = s,n(FaxStop),NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = s,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = s,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = s,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = s,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = s,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = s,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = s,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = s,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = s,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = s,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = s,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = s,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
;zoel : Insert to MySQL fax_activity
exten = s,n,MySQL(Connect connid localhost asterisk)
exten = s,n,GotoIf($[${MYSQL_STATUS} = 0]?:mysql_error,1)
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO fax_activity SET
date=now(),faxfile=${FAXOPT(filename)},faxtxnumber=${FAXTXNUMBER},faxresult=${FAXOPT(status)},faxpages=${FAXOPT(pages)},faxstatus=${FAXOPT(error)},faxline=${CHANNEL})
exten = s,n,MYSQL(Disconnect ${connid})
;zoel : Insert to MySQL fax_activity
exten = mysql_error,1,Noop(Error Connection Mysql)
exten = mysql_error,n,Macro(hangupcall)
I know the dialplan above is not working cause it keep hangup after SendFax for
the 1st time, and it just stop.
So is there any of you guys know how to fix this or is there any otherway so I
can achieve it ?
Thanks
Regards,
ZH
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Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt
Am 19.12.11 14:26, schrieb Zoel Hairi:
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when sending Fax
are very Low especially when we send the Fax just once. Now I’m trying to
modify the dialplan so it will keep trying to send the fax for maximum 5
times at once and it only retry if the Sending Status has Error in it.
Here is the dialplan :
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here … L
hello,
you dont have RetryAttempt2 in your dialplan maybe thats why it stops.
best regards
Stefan
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Re: [asterisk-users] No rtpmap codec info in 200 OK
On 12/19/2011 07:15 AM, William Scott wrote: Ok. Asterisk sends the rtpmap info for the codec. Is it possible to remove this from the 200 OK sent by Asterisk? Possible direction I should look. No, it is not, at least not without patching the Asterisk source code (which of course you are free to do). It seems quite unlikely that the presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any problems. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Sending Fax Dialplan with Retry Attempt
sorry. i put the wrong the dialplan. it already RetryAttempt2 in it.
exten = s,n(RetryAttempt3),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 2 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 3
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt3)
Thanks
ZH
Sent from Samsung Mobile
Stefan Schmidt s...@sil.at wrote:
Am 19.12.11 14:26, schrieb Zoel Hairi:
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when sending Fax
are very Low especially when we send the Fax just once. Now I’m trying to
modify the dialplan so it will keep trying to send the fax for maximum 5
times at once and it only retry if the Sending Status has Error in it.
Here is the dialplan :
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here … L
hello,
you dont have RetryAttempt2 in your dialplan maybe thats why it stops.
best regards
Stefan
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Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Monday, December 19, 2011 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt
Am 19.12.11 14:26, schrieb Zoel Hairi:
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when
sending Fax are very Low especially when we send the Fax just once. Now
I’m trying to modify the dialplan so it will keep trying to send the
fax for maximum 5 times at once and it only retry if the Sending Status
has Error in it.
Here is the dialplan :
exten = s,n,GotoIf($[${FAXOPT(error)} =
NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here … L
hello,
you dont have RetryAttempt2 in your dialplan maybe thats why it stops.
best regards
Stefan
Hi Stefan,
Sorry, My Mistake, in dial plan there's already RetryAttempt2
Below is the Correct One :
[fax-tx]
;Fax For Asterisk - Digium
exten = s,1,NoOp( SENDING FAX )
exten = s,n,Wait(6)
;zoel : Insert to MySQL
exten = s,n,MySQL(Connect connid localhost asterisk)
exten = s,n,GotoIf($[${MYSQL_STATUS} = 0]?:mysql_error,1)
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO fax_activity SET
date=now(),faxfile=${FAXFILE},faxtxnumber=${FAXTXNUMBER},faxresult=Sent to
Spooler,faxpages=${FAXPAGES},faxstatus=${FAXERROR},faxline=${CHANNEL})
exten = s,n,MYSQL(Disconnect ${connid})
;zoel : End Insert to MySQL
; Set FAXOPTs
exten = s,n,NoOp( SETTING FAXOPT )
exten = s,n,Set(FAXOPT(ecm)=yes)
exten = s,n,Set(FAXOPT(headerinfo)=TEST FAX)
exten = s,n,Set(FAXOPT(localstationid)=1234567)
exten = s,n,Set(FAXOPT(maxrate)=14400)
exten = s,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 1 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 2
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here …
exten = s,n(RetryAttempt2),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 2 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 3
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt3)
exten = s,n(RetryAttempt3),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 3 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 4
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt4)
exten = s,n(RetryAttempt4),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 4 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 5
exten = s,n,GotoIf($[${FAXOPT(error)} = NO_ERROR]?FaxStop:RetryAttempt5)
exten = s,n(RetryAttempt5),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 5 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
; Hangup! Print FAXOPTs
exten = s,n(FaxStop),NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = s,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = s,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = s,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = s,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = s,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = s,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = s,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = s,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = s,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = s,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = s,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = s,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
;zoel : Insert to MySQL fax_activity
exten = s,n,MySQL(Connect connid localhost asterisk)
exten = s,n,GotoIf($[${MYSQL_STATUS} = 0]?:mysql_error,1)
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO fax_activity SET
date=now(),faxfile=${FAXOPT(filename)},faxtxnumber=${FAXTXNUMBER},faxresult=${FAXOPT(status)},faxpages=${FAXOPT(pages)},faxstatus=${FAXOPT(error)},faxline=${CHANNEL})
exten = s,n,MYSQL(Disconnect ${connid})
;zoel : Insert to MySQL fax_activity
exten = mysql_error,1,Noop(Error Connection Mysql)
exten = mysql_error,n,Macro(hangupcall)
So do you have any suggestion ?
Thanks
ZH
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[asterisk-users] Dahdi 2.5.0.2 - Strange Warning
Hi, On a recently updated system , I'm now reading lines as this one (never noticed them before): [Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel: Ring requested on unconfigured channel 0/0 span 4 My setup is: Asterisk 1.6.1.18 Libpri 1.4.12 Dahadi 2.5.0.2 My card is a Junghanns QuadBRI. Span 4 config is: ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 group=1,14 context=remote switchtype = euroisdn signalling = bri_cpe_ptmp channel = 10-11 What does it mean ? Should I care ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy in whisper mode - low quality audio
Hi all,
I succeed in injecting audio into one channel by mean of ChanSpy, but this
audio cannot be listened correctly. I am using softphones for Android and
iPhone and there are so many cuts so that they cannot understand what is said
in the audio file. Is this because the total RTP bandwidth is too much?. Which
codec and format should I use for the files played with Playback application?
Dialplan is like this:
same = n,Dial(SIP/${TRUNK}/${ARG2}${NUM},60,M(inject^${CALLERNUMBER}))
[macro-inject]
same = s,1,Originate(Local/trunk@injectWarning,app,Playback,Message-Callee)
same = n,Originate(Local/${ARG1}@injectWarning,app,Playback,Message-Caller)
[injectWarning]
exten = _+34[69],1,Answer()
same = n, ChanSpy(SIP/${EXTEN},qw)
same = n, Hangup()
exten =trunk,1,Answer()
same = n, ChanSpy(SIP/${TRUNK},qw)
same = n, Hangup()
Could anyone help?
Regards,
Isabel
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[asterisk-users] Which Dahdi/Libpri version are you using ?
Hi, I've recently met weird behaviour on 2 different and newly upgraded libpri1.4.12/2dahdi2.5 systems (at the moment, I can't correctly describe the symptoms but that's another story). For various reasons, this lead me to wonder which Dahdi/Libpri combination/version is the most widely used on this planet. Regards -- _ -- 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] Dahdi 2.5.0.2 - Strange Warning
On a recently updated system , I'm now reading lines as this one (never noticed them before): [Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel: Ring requested on unconfigured channel 0/0 span 4 My setup is: Asterisk 1.6.1.18 Libpri 1.4.12 Dahadi 2.5.0.2 My card is a Junghanns QuadBRI. Span 4 config is: ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 group=1,14 context=remote switchtype = euroisdn signalling = bri_cpe_ptmp channel = 10-11 What does it mean ? Should I care ? You could be receiving a signaling only call. The output of pri set debug on span x for the SETUP message is needed to figure out why chan_dahdi/libpri thinks that channel 0 is requested. Richard -- _ -- 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] A lot of 603 Declined Error form iCall - Are they going down or is it just bad service?
Hi everyone, Since three weeks ago, we have been getting A LOT of 603 Declined calls from iCall. I called a few times and their support is either non-responsive (they never call back) or can't fix the issue. I am wondering if everyone else is experiencing the same thing or is it because we recently upgraded from Asterisk 1.6x to Asterisk 1.8x and there is something that is causing this. This happens to their domestic and international routes. I would appreciate the input from those who use their services. Thanks, -- _ -- 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] India Telecom regulations
Hi All, Because I am pretty sure we have people in this DL from India, I was hoping to get the 100% accurate information, is it legal to make calls from any coutry to Indian mobile phones through an Asterisk server based in India? -- Khalid Touati Network Administrator -- _ -- 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 1.6.2.22 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.22. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013: * The sample file listed *two* values for the 'nat' option as being the default. Only 'yes' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22 Thank you for your continued support of Asterisk! -- _ -- 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 1.6.2.22 Now Available
On 12/19/2011 4:08 PM, Asterisk Development Team wrote: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22 or for the non-404-version: http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ChangeLog-1.6.2.22 ;p -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] India Telecom regulations
SIP in India is illegal. Nick. On Mon, Dec 19, 2011 at 3:06 PM, khalid touati khalidtou...@gmail.com wrote: Hi All, Because I am pretty sure we have people in this DL from India, I was hoping to get the 100% accurate information, is it legal to make calls from any coutry to Indian mobile phones through an Asterisk server based in India? -- Khalid Touati Network Administrator -- _ -- 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] India Telecom regulations
On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Use different local IP for each SIP trunk
Hello, I have a SIP provider whom I may want to have multiple trunks with, rather than just adding more channels to the individual trunk. I have discussed the matter with them they have told me that the only way that they identify which trunk should be used for each call is simply by the source IP address that the SIP calls are originating from. They do not use sip username/password or any other means to authenticate the remote caller. With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is there a way in asterisk that I could configure it to use one local IP for the source in all SIP/RTP traffic for 1 SIP trunk then a different local IP for the other SIP trunk? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.comhttp://www.impalanetworks.com/ P: (505) 327-7300 F: (505) 327-7545 -- _ -- 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] No rtpmap codec info in 200 OK
It seems quite unlikely that the presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any problems. Thanks for the reply. I'll expand on the scenario... This particular ATA does not send 'a=rtpmap' for any codec. When talking to a Asterisk PBX everything works fine. When talking to a VSP that sends an INVITE with User-Agent: Sippy the call is setup then drops after 32 seconds. Packet captures shows that no ACK is received after the ATA sends the 200 OK (missing rtpmap). After sending 200 OK about 6 times it then sends BYE and the call disconnects. Every other ATA I have sends rtpmap and works fine. The idea was to manipulate Asterisk into not sending rtpmap for the codec to confirm what happens. I'll now look for another solution. -- _ -- 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] Limit # of inbound calls on SIP trunk
Hello all, I have a system with FreePBX, and as far as I can tell it does not provide a means to limit the number of simultaneous inbound calls on a SIP trunk. Therefore I suspect that I'll need to do some manual dialplan manipulation. Essentially I will have 1 (or possibly 2) SIP trunk(s) configured with a total of 4 channels. Let's say that I only want to permit 2 concurrent inbound calls at any time. If any additional calls hit the asterisk box, I want to play them a busy signal disconnect them. The idea is that I'd like to always have at least 2 channels available for outbound calling at all times. What is the way that you'd recommend accomplishing this? Ultimately, I think that all that I need is a piece of logic in the dialplan when the call initially comes in to say: Are there already 2 existing inbound calls on this SIP trunk? If so, go to context too-many-calls-play-busy. If not, proceed to normal handling. Something like that. Any ideas? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.comhttp://www.impalanetworks.com/ P: (505) 327-7300 F: (505) 327-7545 -- _ -- 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] Limit # of inbound calls on SIP trunk
On Mon, 19 Dec 2011, Douglas Mortensen wrote: I have a system with FreePBX, and as far as I can tell it does not provide a means to limit the number of simultaneous inbound calls on a SIP trunk. Therefore I suspect that I’ll need to do some manual dialplan manipulation. The GROUP() and GROUP_COUNT() functions and the GOTOIF() application should do the trick. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] No rtpmap codec info in 200 OK
I could be wrong but this sounds like a NAT issue rather SIP related packet issue. You are not receiving a response back is what I get a lot of times when my NAT is not setup properly. Call goes on for 10 or 20 second (I try the echo application and it hangs up before I get to talk) and then cuts off. -Bruce On Mon, Dec 19, 2011 at 7:41 PM, William Scott will...@magicwilly.infowrote: It seems quite unlikely that the presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any problems. Thanks for the reply. I'll expand on the scenario... This particular ATA does not send 'a=rtpmap' for any codec. When talking to a Asterisk PBX everything works fine. When talking to a VSP that sends an INVITE with User-Agent: Sippy the call is setup then drops after 32 seconds. Packet captures shows that no ACK is received after the ATA sends the 200 OK (missing rtpmap). After sending 200 OK about 6 times it then sends BYE and the call disconnects. Every other ATA I have sends rtpmap and works fine. The idea was to manipulate Asterisk into not sending rtpmap for the codec to confirm what happens. I'll now look for another solution. -- _ -- 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] No rtpmap codec info in 200 OK
On 20 December 2011 12:51, Bruce B bruceb...@gmail.com wrote: I could be wrong but this sounds like a NAT issue rather SIP related packet issue. I looked at this to start with. Spent sometime comparing addresses and ports between successful and failure packets. Couldn't see any ports that weren't opened on the way out or the use of private ip addresses. I cleared the nat translation table between tests. This ATA works fine with Asterisk based VSPs. I'm just going to have to get more methodical. FYI, the ATA is a GW211 (mass produced OEM device, this one labelled Cormain) and the VSP is Pennytel here in Australia. -- _ -- 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] India Telecom regulations
On Tuesday 20 Dec 2011, Steve Edwards wrote: On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? The only thing that is not permitted is bridging Internet calls with the Indian PSTN. In fact, that too is allowed if you have a VoIP licence from the government. Apart from that, as long as you continue using it within your own organisation, any protocol is fine. IANAL. TINLA. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] India Telecom regulations
Right check out Cordia.LT Sent from my iPhone 4S On Dec 19, 2011, at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: On Tuesday 20 Dec 2011, Steve Edwards wrote: On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? The only thing that is not permitted is bridging Internet calls with the Indian PSTN. In fact, that too is allowed if you have a VoIP licence from the government. Apart from that, as long as you continue using it within your own organisation, any protocol is fine. IANAL. TINLA. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] No rtpmap codec info in 200 OK
Can you register with Eyebeam to VSP and have it work? Make sure you are on the exact same network as the ATA when making this test. This should isolate the NAT issue. On Mon, Dec 19, 2011 at 9:27 PM, William Scott will...@magicwilly.infowrote: On 20 December 2011 12:51, Bruce B bruceb...@gmail.com wrote: I could be wrong but this sounds like a NAT issue rather SIP related packet issue. I looked at this to start with. Spent sometime comparing addresses and ports between successful and failure packets. Couldn't see any ports that weren't opened on the way out or the use of private ip addresses. I cleared the nat translation table between tests. This ATA works fine with Asterisk based VSPs. I'm just going to have to get more methodical. FYI, the ATA is a GW211 (mass produced OEM device, this one labelled Cormain) and the VSP is Pennytel here in Australia. -- _ -- 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] India Telecom regulations
Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) On Mon, Dec 19, 2011 at 10:03 PM, Robert-IPhone rhuddles...@gmail.comwrote: Right check out Cordia.LT Sent from my iPhone 4S On Dec 19, 2011, at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: On Tuesday 20 Dec 2011, Steve Edwards wrote: On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? The only thing that is not permitted is bridging Internet calls with the Indian PSTN. In fact, that too is allowed if you have a VoIP licence from the government. Apart from that, as long as you continue using it within your own organisation, any protocol is fine. IANAL. TINLA. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] India Telecom regulations
On Tuesday 20 Dec 2011, khalid touati wrote: Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) If your objective is to originate calls in the US (using whatever technology), route them over SIP and then terminate them to the PSTN in India, then yes: your Indian presence would need a VoIP licence. Similarly for the reverse: originate a call from Indian PSTN to your local office here and route it using VoIP to any destination (whether within India or abroad). A licence is required in that case too. In general, interconnection of two different entities by bridging Indian PSTN with any other technology requires a licence. If you're only doing VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside India then it's permitted in principle. This is why, e.g., Skype is permitted: it doesn't connect to the Indian PSTN at any stage. Once again, IANAL and TINLA. This is purely from my (mostly informed) understanding of the current laws. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] No rtpmap codec info in 200 OK
On 20 December 2011 14:22, Bruce B bruceb...@gmail.com wrote: Can you register with Eyebeam to VSP and have it work? Make sure you are on the exact same network as the ATA when making this test. This should isolate the NAT issue. Great tip. Eyebeam dosen't send a rtpmap for known codecs unless you select the option too. Well, without it Eyebeam works fine so I better start looking at the firewall. Strange that this particular ATA fails with this particulat VSP only with three different firewalls... vyatta, microtik and a Billion modem/router. Thanks again. -- _ -- 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] Use different local IP for each SIP trunk
AFAIK you can add exterin= in sip.conf for each trunk, correct me if i'm wrong. 2011/12/20 Douglas Mortensen d...@impalanetworks.com Hello, ** ** I have a SIP provider whom I may want to have multiple trunks with, rather than just adding more channels to the individual trunk. I have discussed the matter with them they have told me that the only way that they identify which trunk should be used for each call is simply by the source IP address that the SIP calls are originating from. They do not use sip username/password or any other means to authenticate the remote caller.*** * ** ** With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is there a way in asterisk that I could configure it to use one local IP for the source in all SIP/RTP traffic for 1 SIP trunk then a different local IP for the other SIP trunk? ** ** Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 ** ** -- _ -- 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] Sending Fax Dialplan with Retry Attempt
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Zoel Hairi
Sent: Monday, December 19, 2011 11:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Monday, December 19, 2011 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt
Am 19.12.11 14:26, schrieb Zoel Hairi:
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when
sending Fax are very Low especially when we send the Fax just once.
Now
I’m trying to modify the dialplan so it will keep trying to send the
fax for maximum 5 times at once and it only retry if the Sending
Status
has Error in it.
Here is the dialplan :
exten = s,n,GotoIf($[${FAXOPT(error)} =
NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here … L
hello,
you dont have RetryAttempt2 in your dialplan maybe thats why it
stops.
best regards
Stefan
Hi Stefan,
Sorry, My Mistake, in dial plan there's already RetryAttempt2
Below is the Correct One :
[fax-tx]
;Fax For Asterisk - Digium
exten = s,1,NoOp( SENDING FAX )
exten = s,n,Wait(6)
;zoel : Insert to MySQL
exten = s,n,MySQL(Connect connid localhost asterisk)
exten = s,n,GotoIf($[${MYSQL_STATUS} = 0]?:mysql_error,1)
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO fax_activity
SET
date=now(),faxfile=${FAXFILE},faxtxnumber=${FAXTXNUMBER},faxresult=
Sent to
Spooler,faxpages=${FAXPAGES},faxstatus=${FAXERROR},faxline=${CHAN
NEL})
exten = s,n,MYSQL(Disconnect ${connid})
;zoel : End Insert to MySQL
; Set FAXOPTs
exten = s,n,NoOp( SETTING FAXOPT )
exten = s,n,Set(FAXOPT(ecm)=yes)
exten = s,n,Set(FAXOPT(headerinfo)=TEST FAX)
exten = s,n,Set(FAXOPT(localstationid)=1234567)
exten = s,n,Set(FAXOPT(maxrate)=14400)
exten = s,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 1 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 2
exten = s,n,GotoIf($[${FAXOPT(error)} =
NO_ERROR]?FaxStop:RetryAttempt2)
;= The Call Stop Here …
exten = s,n(RetryAttempt2),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 2 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 3
exten = s,n,GotoIf($[${FAXOPT(error)} =
NO_ERROR]?FaxStop:RetryAttempt3)
exten = s,n(RetryAttempt3),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 3 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 4
exten = s,n,GotoIf($[${FAXOPT(error)} =
NO_ERROR]?FaxStop:RetryAttempt4)
exten = s,n(RetryAttempt4),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 4 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 5
exten = s,n,GotoIf($[${FAXOPT(error)} =
NO_ERROR]?FaxStop:RetryAttempt5)
exten = s,n(RetryAttempt5),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 5 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
; Hangup! Print FAXOPTs
exten = s,n(FaxStop),NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = s,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = s,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = s,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = s,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = s,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = s,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = s,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = s,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = s,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = s,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = s,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = s,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
;zoel : Insert to MySQL fax_activity
exten = s,n,MySQL(Connect connid localhost asterisk)
exten = s,n,GotoIf($[${MYSQL_STATUS} = 0]?:mysql_error,1)
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO fax_activity
SET
date=now(),faxfile=${FAXOPT(filename)},faxtxnumber=${FAXTXNUMBER},f
axresult=${FAXOPT(status)},faxpages=${FAXOPT(pages)},faxstatus=${F
AXOPT(error)},faxline=${CHANNEL})
exten = s,n,MYSQL(Disconnect ${connid})
;zoel : Insert to MySQL fax_activity
exten = mysql_error,1,Noop(Error Connection Mysql)
exten = mysql_error,n,Macro(hangupcall)
So do you have any suggestion ?
Thanks
ZH
Hi All,
So it seems that you cannot use the same channel to SendFax twice in one Dial
Plan at a time,
Re: [asterisk-users] Use different local IP for each SIP trunk
May I ask why do you need different IP addresses to source calls? I mean, its not a common practice, would like to understand the idea behind it. *José Pablo Méndez * On Mon, Dec 19, 2011 at 11:07 PM, Anton Kvashenkin anton.juga...@gmail.comwrote: AFAIK you can add exterin= in sip.conf for each trunk, correct me if i'm wrong. 2011/12/20 Douglas Mortensen d...@impalanetworks.com Hello, ** ** I have a SIP provider whom I may want to have multiple trunks with, rather than just adding more channels to the individual trunk. I have discussed the matter with them they have told me that the only way that they identify which trunk should be used for each call is simply by the source IP address that the SIP calls are originating from. They do not use sip username/password or any other means to authenticate the remote caller. ** ** With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is there a way in asterisk that I could configure it to use one local IP for the source in all SIP/RTP traffic for 1 SIP trunk then a different local IP for the other SIP trunk? ** ** Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 ** ** -- _ -- 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
[asterisk-users] PITCH_SHIFT()
This list is a great resource and I thank all the Asterisk Guru's who actively contribute to it. In Leif Madsen's AstriCon 2010 talk titled 5 Things You Didn't Know Asterisk Could Dohttps://docs.google.com/viewer?a=vq=cache:hCDfIk4pvngJ:leifmadsen.com/sites/default/files/AstriCon%25202010%2520-%25205%2520Things%2520You%2520Didn't%2520Know%2520Asterisk%2520Could%2520Do.pdf+asterisk+dialplan+func+PITCH+SHIFThl=engl=uspid=blsrcid=ADGEEShzSRqJl26lEybK-TvxHL4hKQrd-mBpAapRV6eyI8ST0E5AosCEqp2bm_h5eORZFwwEZDqzEKpT9Fg244nkCgX4BDEGL6bik4Non5_fgm62fzrBxyIXjm1hnqJx2-yGyVlbdXKdsig=AHIEtbQ2NyYajUzeJshmWKAgZEi0RprNjQpli=1 he mentions that the PITCH_SHIFT() function is designed to be used dynamically and can change the pitch of a channel on the fly using features.conf. Can someone provide me with any information of how this would be accomplished for dynamic use? I'm familiar with the dialplan syntax use examples such as: exten = 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave exten = 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more exten = 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch and so forth, but don't understand how these functions would be called dynamically from features.conf. any help is greatly appreciated... -- _ -- 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
