Re: [asterisk-users] Interesting email project.
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@xxx.com. with the
time date and CID included in the email. I know how to code some but am
looking for the best way to do this.
something like this?
exten = _713X.,1,System(/web/html/icq.php censored UIN [VoIP]
Incoming call, CLID: ${CALLERID(num)}.)
exten = _713X.,n,System(echo Incoming call at `date`. | /bin/mail -s
InCall from ${CALLERID(num)} at `date`. censored e-mail)
regards
adam
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Re: [asterisk-users] Channel failover
On 4 May 2010, at 03:44, Jack Bates wrote: We recently got VoIP, so when we make a call, Asterisk should first try to make the call with VoIP, but in case either our VoIP or our internet service are down, Asterisk should then try to make the call with our old school analog phone line Well, first you try to dial it with the VoIP line.. Then the analogue one... So you just put the two dial commands on separate lines.. S -- _ -- 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 and Patton
On Fri, 30 Apr 2010 18:52:46 +0200, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: As I said, you could think about creating 4 different SIP gateways on the Patton with 4 differing SIP ports. I don't know if the Patton will handle 4 gateways - but it might. We have 4 trunk and 4 company in our office, I was testing FOP and I would want to show the occupied trunks for inbound and outbound calls for single company. Alternatives are: - use GROUP() and GROUP_COUNT in the dialplan - use DEVICE_STATE in the dialplan This includes a lenghty example on how to monitor a BRI trunk: http://www.voip-info.org/wiki/view/Asterisk+func+device_State Philipp Hi Philipp, thanks, for your help I'll try to find a solution to use FOP. Best regards. Eco. -- _ -- 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] DSCP QoS value in YeaLink phone settings
Hello list, I need to set Voice QoS and SIP QoS for YeaLink. The possible values are 0 ~ 63. With Grandstream I can fill in DiffServ 46, which is EF. That's what I want. With Snom I fill in 184, which corresponds to EF or DSCP 46 (according to their wiki) But what value do I want to fill in with this YeaLink ??? This is a conversion table : http://www.cnetds.com/docs/DSCP-ToS-AF-binary-conversionTable.html I think I'm looking for 184, but YeaLink does not let me fill in a number 63... Thank you for your help. Jonas. -- _ -- 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] Check if extension loaded over AMI
Hello list, I was wondering if there is a way to see if a given piece of dialplan is loaded through AMI. I have seen the GetConfig command, but it seems to expect a file name to retrieve, and I don't necessarily know that (as it could be down the line bu multiple levels of #includes from the main extensions.conf). I could run an AMI Command to run the cli command dialplan show mycontext, but I'm a bit worried by the performance cost of running a non-natively AMI command; plus I don't love much the line-formatted response. I could create a dummy piece of dialplan that is in the same place as . the one i want to check, and I could try and Originate that and see if it found or not. All solutions above seem to be suboptimal any idea? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] Reading the CDR
Dan Journo wrote: - you could also consider the M() option to Dial together with the CDR userfield for logging whatever channel variable make sense to you I'll see if I can sort it out with that. - have you looked at the destination channel in the CDR? The destination channel says:- SIP/sipprovider-002c The 002c increments by one for each new call. I've had this same issue. I don't know if this will help in your situation but we process all our CDR into a different DB and we also have the CDR provided by the sip provider at hand as well so the processing script checks to see what number an outgoing call was made to at that time in the CDR of the sip provider. A bit long winded but seeing as we're already doing a lot of processing it's just an extra check to add in certain circumstances. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 T.38 fax negotiation problem
Hi, I'm experiencing the same problem with t38modem and hylafax. My problem is that on the re-Invite phase it syncs lower than 2400 bpps and the connection hangs on the second page. Could you please post here the patch for asterisk 1.6.2.4 or even indicate which is the trunk of asterisk where this patch take effect? thanks a lot, Miguel Amez. 2010/5/3 Kevin P. Fleming kpflem...@digium.com On 05/03/2010 11:59 AM, Ilmars Knipshis wrote: Problem in short is as following: after reINVITE from Cisco to negotiate T.38: --- SIP read from UDP:193.110.9.17:5060 --- INVITE sip:37166101...@159.148.78.220 sip%3a37166101...@159.148.78.220SIP/2.0 Via: SIP/2.0/UDP 193.110.9.17:5060 From: sip:3250890...@193.110.9.17 sip%3a3250890...@193.110.9.17 ;tag=74ff1200077fff10ff18ff29ff16 To: 3716610 sip:37166101...@159.148.78.220sip%3a37166101...@159.148.78.220 ;tag=as32fabaec Call-ID: 46ba3dad03495f6f3542698033470...@159.148.78.220 CSeq: 103 INVITE Contact: sip:3250890...@193.110.9.17 sip%3a3250890...@193.110.9.17 ;user=phone Max-Forwards: 10 User-Agent: MERA MSIP v.1.0.2 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1272610573 1272610573 IN IP4 193.110.9.17 s=- c=IN IP4 193.110.9.17 t=0 0 m=image 25296 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy The problem here is being caused by the re-INVITE occurring prior to SendFAX() being started; this really should not be happening, as the other endpoint should not re-INVITE until it knows that a FAX endpoint is calling, but some of them do this anyway. There is a fix for this problem in SVN Asterisk trunk already, and it will be merged into the 1.6.2 branch in the next couple of weeks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- _ -- 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] working example of t38 fax w/ 1.6.2?
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on the
better way.
regards,
Miguel Amez
2010/5/4 sean darcy seandar...@gmail.com
Miguel Amez wrote:
Hi Sean,
Do you know about t38modem and hylafax?
There are lots of wonderfull options with both of them.
If you need config files with both of them tell me.
See ya
2010/5/2 sean darcy seandar...@gmail.com mailto:seandar...@gmail.com
I can't get a test T.38 fax between 2 1.6.2 machines, using app
_fax and spandsp pre17 and 20100501. The machines can't seem to get
connected.
send side extensions.conf:
[fax-tx-test]
exten=s,1,NoOp(Context fax-tx-test)
exten=s,n,SendFAX(${FaxFile}.tif)
exten=s,n,HangUp()
exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR}
FAXMODE:
${FAXMODE})
Channel:SIP/side-sip-fax
Context:fax-tx-test
Extension:s
Priority:1
Set:FaxFile=/var/spool/asterisk/fax/20091113_1455
receive side:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif)
exten = s,n,Hangup()
There's a bunch more stuff at
https://issues.asterisk.org/view.php?id=17105
But does anyone have a setup that Just Works? I'd love to find a
setup
that works for someone else and just copy it.
Thanks,
sean
Yes, I am familiar with Hylafax. But I'm trying to Keep It Simple, and
just use app_fax. Is it working for anyone? Does anybody have a simple
working example?
sean
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Re: [asterisk-users] sending T.38 fax negotiation problem
On 05/04/2010 06:30 AM, Miguel Amez wrote: I'm experiencing the same problem with t38modem and hylafax. My problem is that on the re-Invite phase it syncs lower than 2400 bpps and the connection hangs on the second page. The patch I'm talking about won't affect t38modem and Hylafax usage at all. If the re-INVITE arrives before you have connected the call to t38modem, the negotiation process will very likely fail. Could you please post here the patch for asterisk 1.6.2.4 or even indicate which is the trunk of asterisk where this patch take effect? The patch for 1.6.2 is being tested today and should be merged before the end of the day. It won't be for 1.6.2.4, because that's not the current release, but will be against 1.6.2.7. It will be in a 1.6.2 release candidate in a few days. Once the patch is in 1.6.2, we'll release a new version of Fax For Asterisk that can take advantage of it... hopefully also this week. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 T.38 fax negotiation problem
Kevin P. Fleming wrote: On 05/04/2010 06:30 AM, Miguel Amez wrote: I'm experiencing the same problem with t38modem and hylafax. My problem is that on the re-Invite phase it syncs lower than 2400 bpps and the connection hangs on the second page. The patch I'm talking about won't affect t38modem and Hylafax usage at all. If the re-INVITE arrives before you have connected the call to t38modem, the negotiation process will very likely fail. Typically HylaFAX users have the calls connected to the modems from the outset. Thanks, Lee. -- _ -- 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] Interesting email project.
wow thanks guys. Ill try it out.
Respectfully
Michael D Mosier
Ftoc Certified
On May 4, 2010 1:36 AM, ad...@3a.hu wrote:
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@x...
something like this?
exten = _713X.,1,System(/web/html/icq.php censored UIN [VoIP]
Incoming call, CLID: ${CALLERID(num)}.)
exten = _713X.,n,System(echo Incoming call at `date`. | /bin/mail -s
InCall from ${CALLERID(num)} at `date`. censored e-mail)
regards
adam
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Re: [asterisk-users] Interesting email project.
whats censored UIN? [VoIP]
On Tue, May 4, 2010 at 8:00 AM, mike mosier trixbo...@gmail.com wrote:
wow thanks guys. Ill try it out.
Respectfully
Michael D Mosier
Ftoc Certified
On May 4, 2010 1:36 AM, ad...@3a.hu wrote:
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@x...
something like this?
exten = _713X.,1,System(/web/html/icq.php censored UIN [VoIP]
Incoming call, CLID: ${CALLERID(num)}.)
exten = _713X.,n,System(echo Incoming call at `date`. | /bin/mail -s
InCall from ${CALLERID(num)} at `date`. censored e-mail)
regards
adam
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[asterisk-users] Problem with AMI Originate
Hello there, How to retrieve the failure reason when calling AMI Originate with Async = 0? The system seems to return the following no matter what: [Response] = Error [Message] = Originate failed How to determine if the number was busy, invalid, etc? Would I have to run AMI Originate with Async = 1 in order to do that? Thanks in advance, Leo PS. I'm running Asterisk 1.6.2.0~rc2 -- _ -- 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] Interesting email project.
mike mosier wrote:
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com
mailto:mmos...@xxx.com. with the time date and CID included in the
email. I know how to code some but am looking for the best way to do this.
Sorry I might have asked this a couple months back. I forgot.
Mmosier
Houston
Respectfully
Michael D Mosier
Ftoc Certified
Here is the script I am using for email alert. Form Asterisk dialplan:
exten = h,1,System(/path/to/the/script/emailnotice.sh
some...@gmail.com ${CALLERID(num)} ${CALLERID(name)}
${DIALSTATUS} ${VMSTATUS} ${MYEXTEN}
${STRFTIME(${EPOCH},,%Y/%m/%d %H:%M)})
==
#!/bin/sh
#Used for email alerting of incoming calls
#$1 email address
#$2 callerid num
#$3 callerid name
#$4 dial status
#$5 vm status
#$6 extension
#$7 datetime
if [ $# != 7 ]
then exit 1
fi
#Store command line args in nice variables
EMAIL=$1
CALLERIDNUM=$2
CALLERIDNAME=$3
DIALSTATUS=$4
VMSTATUS=$5
EXTEN=$6
TIMESTAMP=$7
DEBUG=0 #Set to 0 for standard operation. 1 will log inputs and
mail commands for debugging.
LOGFILE=/var/log/asterisk/IncomingCalls.log
SENDMAIL=1 #Set to 1 if you want it to email the alert. 0 is
useful for debugging.
EMAILCMD=/usr/sbin/sendmail -t
SUBJECT=Incoming call to ${EXTEN} at ${TIMESTAMP}
#log mail command
if [ ${DEBUG} -eq 1 ]; then
echo $1 $2 \$3\ $4 $5 $6 $7 ${LOGFILE}
fi
#Check we have an email address if not quit
if [ ${EMAIL} = ]; then
exit 0
fi
if [ ${DIALSTATUS} = CANCEL ]; then
BODY=Caller - ${CALLERIDNAME} (${CALLERIDNUM}) hung up.
elif [ ${DIALSTATUS} = ANSWER ]; then
BODY=Caller - ${CALLERIDNAME} (${CALLERIDNUM}) was answered by
${EXTEN}.
else
BODY=[ ${DIALSTATUS} ] ${CALLERIDNAME} (${CALLERIDNUM}) hung up.
#check for hangup in vm menu. ex call went to vm and user hung up
if [ ${VMSTATUS} = USEREXIT ]; then
BODY=[ ${DIALSTATUS} ] ${CALLERIDNAME} (${CALLERIDNUM})
hung up on vm.
fi
#check for hangup in vm menu. ex call went to vm and user hung up
if [ ${VMSTATUS} = FAILED ]; then
BODY=[ ${DIALSTATUS} ] ${CALLERIDNAME} (${CALLERIDNUM})
hung up on vm.
fi
#if they left a vm we already would get an email. Don't need a 2nd
if [ ${VMSTATUS} = SUCCESS ]; then
exit 0
fi
fi
#log mail command
if [ ${DEBUG} -eq 1 ]; then
echo To: ${EMAIL} Subject: ${SUBJECT} ${BODY} ${LOGFILE}
fi
#send email
if [ ${SENDMAIL} -eq 1 ]; then
${EMAILCMD} INLINE
From: aster...@mydomain.com
To: ${EMAIL}
Subject: ${SUBJECT}
${BODY}
INLINE
fi
exit 0
--
Jian Gao
IT Technician
SJ Geophysics Ltd. http://www.sjgeophysics.com
jian@sjgeophysics.com mailto:jian@sjgeophysics.com
Tel: (604)582-1100
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[asterisk-users] queue members
Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via -QueueAdd(4050, Local/4...@from-internal/n, 1) (not via agents.conf). SIP extension 4053 rings, answers and then decides to blind-transfer to ZAP/DAHDI extension 3666. The show queue command still displays 4053 as In use. However, if 3210 calls 4050 and 4053 answers and finally hangs up (no transfer) then the show queue command does not display In use. What's the difference? # asterisk -rx show queue 4050 -- Remote UNIX connection 4050 has 0 calls (max 6) in 'ringall' strategy (1s holdtime), W:0, C:1,A:1, SL:0.0% within 0s Members: Local/4...@from-internal/n with penalty 1 (dynamic) (In use) has taken 1 calls (last was 99 secs ago) No Callers Verbosity is at least 3 # asterisk -rx show channels concise Zap/2-1:from-alcatel-custom:s:1:Up:Bridged Call:Local/4...@from-internal-4120,2:4053::3::Local/4...@from-internal-4120,2 Local/4...@from-internal-4120,2:macro-dialout-trunk:s:19:Up:Dial:ZAP/g1/3666|300|tTwWM(auto-blkvm):3210::3:122:Zap/2-1 Local/4...@from-internal-4120,1:from-internal:s:1:Up:Bridged Call:IAX2/coinbound-1551:3210::3::IAX2/coinbound-1551 IAX2/coinbound-1551:ext-queues:4050:19:Up:Queue:4050|t||:3210::3:128:Local/4...@from-internal-4120,1 Verbosity is at least 3 Thanks, Vieri -- _ -- 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] Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring
phone for 20 seconds
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
2010/4/29 Peder pe...@networkoblivion.com
In PBX1, where are you actually dialing the phone? The first line of
the macro just says “goto dialstatus” with no Dial statement.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, April 29, 2010 2:03 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is
written) but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1)
to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1
it just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1)
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1]
VoiceMail(IAX2/pbx2-15464, u8029) in new stack
*[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
Failed to write frame*
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on
'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or
fix the issue I'm having, thanks a lot!
--
Abdullah
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Abdullah
--
Abdullah
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[asterisk-users] client-server encryption
Hi, I'm trying to set up a secure VoIP channel between a Windows softphone client and an Asterisk 1.6... server running with OpenBSD. By secure I mean to prevent any man in the middle to reconstitute any vocal exchange nor sender/addressee/any header data/ of the VoIP call (in first step, I would be glad to secure vocal data ans see later for the header...) I had a look to several way to do that: - Create a VPN using OpenVPN = impossible for me , i'm not admin of the Windows system. - Create a SSH tunnel from the Windows client to the Asterisk server using putty (redirecting ports used for VoIP) = it doesn't work because either SIP/RTP or IAX2 protocol are based on UDP so that SSH tunneling isn't working - Use IAX2 protocol to communicate (because I was told it was able to encrypt data) = it doesn't work because none of the client I had support encryption (many deal with authentication encryption but not stream data)... Do you know a client which could do that ? Now I tried all of this, I do not have other idea... Do you have any ? Each clue is very welcome! 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
Re: [asterisk-users] client-server encryption
Hi, On 05-04-2010 18:46, isca...@free.fr wrote: - Create a VPN using OpenVPN = impossible for me , i'm not admin of the Windows system. this is a bad thing, but the vpn concept might work after all. have you considered a pptp/l2tp/ipsec vpn? AFAIK on the client side, you may succeed without admin privileges and it's only a matter of pppd/pptpd/l2tpd/*swan on the server side. if the local LAN is trusted, you may deploy a vpn capable device with the purpose of establishing a vpn to the server. it's only a routing issue from there. regards adam -- _ -- 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] Interesting email project.
- Original Message -
mike mosier wrote:
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com
mailto:mmos...@xxx.com. with the time date and CID included in the
email. I know how to code some but am looking for the best way to do
this.
Sorry I might have asked this a couple months back. I forgot.
Mmosier
Houston
Respectfully
Michael D Mosier
Ftoc Certified
Here is the script I am using for email alert. Form Asterisk dialplan:
exten = h,1,System(/path/to/the/script/emailnotice.sh
some...@gmail.com ${CALLERID(num)} ${CALLERID(name)}
${DIALSTATUS} ${VMSTATUS} ${MYEXTEN}
${STRFTIME(${EPOCH},,%Y/%m/%d %H:%M)})
you could always use the PHP AGI interface to send the email and log
information to a database ?
eg.
exten = h,1,AGI(sendemailandlog.php)
--
Thanks, Phil
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[asterisk-users] Asterisk 1.6.0.27 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.27. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes only, scheduled for the first half of May 2010. The Asterisk development team recommends that all users of Asterisk 1.6.0 and 1.6.1 series move to the 1.6.2 series for continued bug fix support. More information about the changes to maintenance support can be found at: http://www.asterisk.org/node/49924 Information about the Asterisk maintenance schedule is available at: http://www.asterisk.org/asterisk-versions The release of Asterisk 1.6.0.27 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix building CDR and CEL SQLite3 modules. (Closes issue #17017. Reported by alephlg. Patched by seanbright) * Resolve a crash in SLAtrunk() when the specified trunk does not exist. (Reported in #asterisk-dev by philipp64. Resolved by seanbright) * Update to new Local channel documentation. (Closes issue #16963. Reported, patched by kobaz) * Make safe_asterisk work on dash/sh/bash, etc. (Closes issue #17094. Reported by stuarth. Tested by pabelanger. Patched by mvanbaak) * Pass the PID of the Asterisk process, not the PID of the canary. (Closes issue #17065. Reported by globalnetinc. Patched by makoto. Tested by frawd, globalnetinc) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.27 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
[asterisk-users] Asterisk 1.6.1.19 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.1.19. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes only, scheduled for the first half of May 2010. The Asterisk development team recommends that all users of Asterisk 1.6.0 and 1.6.1 series move to the 1.6.2 series for continued bug fix support. More information about the changes to maintenance support can be found at: http://www.asterisk.org/node/49924 Information about the Asterisk maintenance schedule is available at: http://www.asterisk.org/asterisk-versions The release of Asterisk 1.6.1.19 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix building CDR and CEL SQLite3 modules. (Closes issue #17017. Reported by alephlg. Patched by seanbright) * Resolve crash in SLAtrunk when the specified trunk doesn't exist. (Reported in #asterisk-dev by philipp64. Patched by seanbright) * Update code to reflect that handle_speechset has 4 arguments. (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, mmichelson) * Pass the PID of the Asterisk process, not the PID of the canary. (Closes issue #17065. Reported by globalnetinc. Patched by makoto. Tested by frawd, globalnetinc) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.19 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
[asterisk-users] Asterisk 1.6.2.7 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix building CDR and CEL SQLite3 modules. (Closes issue #17017. Reported by alephlg. Patched by seanbright) * Resolve crash in SLAtrunk when the specified trunk doesn't exist. (Reported in #asterisk-dev by philipp64. Patched by seanbright) * Include an extra newline after Aliased CLI command to get back the prompt. (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright) * Prevent segfault if bad magic number is encountered. (Issue #17037. Reported, patched by alecdavis) * Update code to reflect that handle_speechset has 4 arguments. (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, mmichelson) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.7 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
[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server - it is robbery!
All, Thanks for the suggestions, but the system is a plan non-sip, non-ip, non pri setup. It's pretty much a closed box setup. And the prices for the card and support are robbery - which is why we aren't going to go with another setup like that. While it has been reliable - I don't think there has ever been an issue with it, expansion is expensive. The local company was gouging us with $200 per incident (ie add an extension) service calls, until I found an installation manual on google, and downloaded. They griped because I was using it, but hey, it wasn't that hard to figure out. So might as well jump off the cliff and go full scale asterisk! Thanks, Eddie -- _ -- 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 1.4.31 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.31. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.31 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved with the help of the community: * Ensure line terminators in email are consistent. (Closes issue #16557. Reported by jcovert. Tested by ebroad, zktech) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) * Resolve a deadlock in chan_local. (Closes issue #17185. Reported by schmoozecom. Tested by schmoozecom, GameGamer43. Patched by dvossel) * Fix crash in audiohook_write_list. (Closes issue #17052, #16196. Reported by dvossel, atis. Patched by dvossel) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.31 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] client-server encryption
Iscario- I'm trying to set up a secure VoIP channel between a Windows softphone client and an Asterisk 1.6... server running with OpenBSD. By secure I mean to prevent any man in the middle to reconstitute any vocal exchange nor sender/addressee/any header data/ of the VoIP call (in first step, I would be glad to secure vocal data ans see later for the header...) I had a look to several way to do that: - Create a VPN using OpenVPN = impossible for me , i'm not admin of the Windows system. - Create a SSH tunnel from the Windows client to the Asterisk server using putty (redirecting ports used for VoIP) = it doesn't work because either SIP/RTP or IAX2 protocol are based on UDP so that SSH tunneling isn't working - Use IAX2 protocol to communicate (because I was told it was able to encrypt data) = it doesn't work because none of the client I had support encryption (many deal with authentication encryption but not stream data)... Do you know a client which could do that ? Now I tried all of this, I do not have other idea... Do you have any ? Each clue is very welcome! Run through Kamailio server + rtpproxy, use SRTP (or other) encryption extension to rtpproxy. -Jeff -- _ -- 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] Code in extensions.conf to leave a voice mailin another PBX ?!
See if this helps
http://www.voipuser.org/forum_topic_3921.html
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Tuesday, May 04, 2010 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Code in extensions.conf to leave a voice
mailin another PBX ?!
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring phone
for 20 seconds
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
2010/4/29 Peder pe...@networkoblivion.com
In PBX1, where are you actually dialing the phone? The first line of the
macro just says goto dialstatus with no Dial statement.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Code in extensions.conf to leave a voice mail in
another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1) in
new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1] VoiceMail(IAX2/pbx2-15464,
u8029) in new stack
[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed
to write frame
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on 'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
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Abdullah
--
Abdullah
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Re: [asterisk-users] sending T.38 fax negotiation problem
Hi Kevin. That sounds marvellous! Maybe some of my problems come from that issue, so tomorrow's revision of 1.6.2.7 could have a solution. Facing another problem, as I told you, I'm experiencing some troubles with t38modem's configuration, and I would like to know if you had have experience with t38modem and hopefully we could share information about configuring this and publish the whole documentation on an internet forum, because previous tries of doing this like these: http://www.voip-info.org/wiki/view/T38modem+configuration+with+Asterisk http://www.foriamroot.org/hylafax-6-0-debian-or-ubuntu-t38modem-1-0-asterisk-1-6/ Have some troubles and compatibility issues. I could give you lots of asterisk logs indeed for the troubles I'm facing with re-Invite. Thanks for the feedback. Regards, Miguel Amez 2010/5/4 Kevin P. Fleming kpflem...@digium.com On 05/04/2010 06:30 AM, Miguel Amez wrote: I'm experiencing the same problem with t38modem and hylafax. My problem is that on the re-Invite phase it syncs lower than 2400 bpps and the connection hangs on the second page. The patch I'm talking about won't affect t38modem and Hylafax usage at all. If the re-INVITE arrives before you have connected the call to t38modem, the negotiation process will very likely fail. Could you please post here the patch for asterisk 1.6.2.4 or even indicate which is the trunk of asterisk where this patch take effect? The patch for 1.6.2 is being tested today and should be merged before the end of the day. It won't be for 1.6.2.4, because that's not the current release, but will be against 1.6.2.7. It will be in a 1.6.2 release candidate in a few days. Once the patch is in 1.6.2, we'll release a new version of Fax For Asterisk that can take advantage of it... hopefully also this week. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- _ -- 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] problem with ringinuse=no, queue members receive randomly two calls
Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the user goes in a status Ring+Inuse. Any idea? Why the call is still dispatched to the user if it is not in the Not in use status? Thanks to all in advance * * LOG (core debug and verbose set to 5) * * #grep PL1038 full [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:08] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 6 (Ringing) [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] VERBOSE[30453] logger.c: -- SIP/PL1038-5f7d is ringing [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:08] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 6 (Ringing) [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] VERBOSE[30268] logger.c: -- SIP/PL1038-5f7e is ringing [May 4 16:21:10] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:10] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:10] DEBUG[3035] chan_sip.c: build_route: Contact hop: sip:pl1...@10.192.37.119 [May 4 16:21:10] DEBUG[30268] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:10] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:10] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:10] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:10] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:10] VERBOSE[30268] logger.c: -- SIP/PL1038-5f7e answered SIP/192.168.55.32-5f59 [May 4 16:21:10] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:14] VERBOSE[30268] logger.c: -- Native bridging SIP/192.168.55.32-5f59 and SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:14] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:14] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:14] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:14] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:15] DEBUG[29938] app_queue.c: Trying 'SIP/PL1038' with metric 0 [May 4 16:21:15] DEBUG[29938] app_queue.c: SIP/PL1038 in use, can't receive call [May 4 16:21:16] DEBUG[30097] app_queue.c: Trying 'SIP/PL1038' with metric 0 [May 4 16:21:16] DEBUG[30097] app_queue.c: SIP/PL1038 in use, can't receive call [ * * config * * sip users: [PL1039] context=mycontext callerid=PhoneLine1039 1039 secret=pwd1039 type=peer host=dynamic call-limit=3 disallow=all allow=ulaw queues: [queue_1] weight=10 wrapuptime=0 strategy=leastrecent joinempty=no retry=0 autopause=yes setinterfacevar=yes eventwhencalled=yes eventmemberstatus=yes ringinuse=no member = SIP/PL1039 [queue_2] weight=10 wrapuptime=0 strategy=leastrecent joinempty=no retry=0 autopause=yes setinterfacevar=yes eventwhencalled=yes eventmemberstatus=yes ringinuse=no member = SIP/PL1039
Re: [asterisk-users] sending T.38 fax negotiation problem
To make it clear, the change was merged to the 1.6.2 branch recently, and will not be in 1.6.2.7 as those releases candidates were made a couple of weeks ago. The changes will be available in the next set of release candidates, slated to be 1.6.2.8-rc1 sometime this week. Leif. Miguel Amez wrote: That sounds marvellous! Maybe some of my problems come from that issue, so tomorrow's revision of 1.6.2.7 could have a 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
Re: [asterisk-users] working example of t38 fax w/ 1.6.2?
On 5/4/2010 7:32 AM, Miguel Amez wrote:
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on
the better way.
regards,
Miguel Amez
2010/5/4 sean darcy seandar...@gmail.com mailto:seandar...@gmail.com
Miguel Amez wrote:
Hi Sean,
Do you know about t38modem and hylafax?
There are lots of wonderfull options with both of them.
If you need config files with both of them tell me.
See ya
2010/5/2 sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com mailto:seandar...@gmail.com
mailto:seandar...@gmail.com
I can't get a test T.38 fax between 2 1.6.2 machines, using app
_fax and spandsp pre17 and 20100501. The machines can't seem
to get
connected.
send side extensions.conf:
[fax-tx-test]
exten=s,1,NoOp(Context fax-tx-test)
exten=s,n,SendFAX(${FaxFile}.tif)
exten=s,n,HangUp()
exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR}
FAXMODE:
${FAXMODE})
Channel:SIP/side-sip-fax
Context:fax-tx-test
Extension:s
Priority:1
Set:FaxFile=/var/spool/asterisk/fax/20091113_1455
receive side:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif)
exten = s,n,Hangup()
There's a bunch more stuff at
https://issues.asterisk.org/view.php?id=17105
But does anyone have a setup that Just Works? I'd love to
find a setup
that works for someone else and just copy it.
Thanks,
sean
Yes, I am familiar with Hylafax. But I'm trying to Keep It Simple, and
just use app_fax. Is it working for anyone? Does anybody have a simple
working example?
sean
It's the fax module built into 1.6.2.
sean
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Re: [asterisk-users] Asterisk 1.6.2.7 Now Available
On 5/4/2010 1:59 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix building CDR and CEL SQLite3 modules. (Closes issue #17017. Reported by alephlg. Patched by seanbright) * Resolve crash in SLAtrunk when the specified trunk doesn't exist. (Reported in #asterisk-dev by philipp64. Patched by seanbright) * Include an extra newline after Aliased CLI command to get back the prompt. (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright) * Prevent segfault if bad magic number is encountered. (Issue #17037. Reported, patched by alecdavis) * Update code to reflect that handle_speechset has 4 arguments. (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, mmichelson) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.7 Thank you for your continued support of Asterisk! If I'm reading the ChangeLog correctly 1.6.2.7 = 1.6.2.7-rc3. Right? sean -- _ -- 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 1.6.2.7 Now Available
sean darcy wrote: If I'm reading the ChangeLog correctly 1.6.2.7 = 1.6.2.7-rc3. Right? Correct -- all releases are a direct copy of the last release candidate (in nearly all cases anyways). Leif. -- _ -- 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] Productivity Suite on Polycom IP7000
Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes' setting that I need to toggle? Also, does anybody know any good therapists with a side-specialty of torn-out hair replacement? :-) Thanks in advance! -Karl -- _ -- 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] Productivity Suite on Polycom IP7000
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Tuesday, May 04, 2010 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Productivity Suite on Polycom IP7000 Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes' setting that I need to toggle? Also, does anybody know any good therapists with a side-specialty of torn-out hair replacement? :-) According to the release notes (I'm looking at 3.2.3), 4-way conferencing is not possible on the IP7000s. In fact, any of the features that are supported that would otherwise require a Productivity License (LDAP, Conference Management) are available without any license. Regards, - Brad -- _ -- 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 1.6.2.7 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. What version of Skype for Asterisk works with this release? -- _ -- 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 1.6.2.7 Now Available
Richard Kenner wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. What version of Skype for Asterisk works with this release? Should be the latest available on the Digium downloads site. It says version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2 branch for quite some time (I just updated it last week). Leif. -- _ -- 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 1.6.2.7 Now Available
Should be the latest available on the Digium downloads site. It says version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2 branch for quite some time (I just updated it last week). Hmm. So was I until it abruptly stopped working. It started again when I went back to an older SVN revision. Maybe I should try 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] Productivity Suite on Polycom IP7000
- Original Message - From: Watkins, Bradley bradley.watk...@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 04, 2010 4:50 PM Subject: Re: [asterisk-users] Productivity Suite on Polycom IP7000 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Tuesday, May 04, 2010 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Productivity Suite on Polycom IP7000 Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes' setting that I need to toggle? Also, does anybody know any good therapists with a side-specialty of torn-out hair replacement? :-) According to the release notes (I'm looking at 3.2.3), 4-way conferencing is not possible on the IP7000s. In fact, any of the features that are supported that would otherwise require a Productivity License (LDAP, Conference Management) are available without any license. Regards, - Brad Thanks Brad. That matches my observation. It seems like such an ironic a feature omission as to be absurd. The expensive _conference_ phone seems (to me) to be precisely the most likely to support conference-like features. We actually first bumped into this 'problem' over a year ago when the phone was first released. We were early adopters. We put in support tickets in with Polycom but got no love. We gave up, only now revisiting it. You'd think someone in support might have known it was not supported way back then. What's most interesting about this is that if you look at the ORIGINAL productivity suite marketing Flash videos, they actually show FIVE-way [sic] conferencing on an IP670. That's five-way, as in YOU and 4 other callers. I suspect that 5-way was originally the hope, but in the real-world it ended up being too resource intensive to provide consistent quality. It seems reasonable that the feature may have simply been 'scaled back' from FIVE to FOUR-way calling on the IP6xx. By extension, perhaps the IP7000 (supporting full 14khz audio) found its 'real world' limit at THREE-way calling instead of FOUR. Just a WAG, but maybe a reasonable one. -Karl -- _ -- 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 1.6.2.7 Now Available
Richard Kenner wrote: Should be the latest available on the Digium downloads site. It says version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2 branch for quite some time (I just updated it last week). Hmm. So was I until it abruptly stopped working. It started again when I went back to an older SVN revision. Maybe I should try again. If that is the case, then you may need to contact Digium support, but as far as I can tell mine is still working with a recent SVN checkout of 1.6.2. *some time goes by* OK, I got sufficiently curious to make sure Skype for Asterisk still loaded on 1.6.2.7. It does for me, but I had to run make install in my Skype source directory. One of the modules loaded, but the 'skype' CLI command was not available until after I ran make install again, so one of the Skype for Asterisk components must not have been compatible. Leif. -- _ -- 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] Problems with Asterisk 1.6.2.1 working in Realtime with PostgreSQL
Hi Anyone, I have a server with asterisk 1.6.2.1 working in Realtime with PostgreSQL, but I'm having problems when happened any error in a table, for example, automatically this error stop the Asterisk. Has a way to configure the DB that when happened any problem don't stop the asterisk? Thank so much. Bye -- _ -- 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] Transfer calls using ##
I have a question about the blind transfer using ##. This works great on our cordless phone, but there have been occasions that we can't transfer using ##. I was able to reproduce the issue by doing the following: 1) Call in from the outside line, 2) Ask the operator to transfer me to an extension using ##. 3) Get the voice mail greeting of the individual. 4) Hit 0 for the operator before the greeting completed. 5) Ask the operator to transfer me again using ##. 6) Operator can't transfer and I can hear the pressing of the keys. Why can't I transfer the call the second time around? How can I fix this? -- _ -- 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
