[asterisk-users] asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift thanks darren!) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip set debug peer vgw1 (vgw1 is my cisco 1760 ata) http://jeremy.kister.net/tmp/fax/console.txt http://jeremy.kister.net/tmp/fax/messages.txt http://jeremy.kister.net/tmp/fax/sip.txt I've tried using the packaged app_fax_spandsp and also Digium's app_fax_digum for 1.8.0-rc1 -- no difference in behavior. Anyone have any ideas how I can get this fixed? -- 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
[asterisk-users] CallerID from Samsung PBX line on FXO
Hi, I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX via an 8SLI analogue extension card in the Samsung and an Openvox A400P04 4-FXO card in the Asterisk box. It all works in that I can place calls in both directions from the office Samsung extensions and Asterisk SIP extensions. The only tricky bit was getting the FXO to detect hang-up from the Samsung correctly - a tweak on the Samsung config to make it hang up properly fixed this. So far Asterisk has been a joy to configure (apart from the default config which IMO should be stripped back to bare-bones and the config shipped as examples in the doc folder - this may be the fault of my distribution who's packages I used rather than get in to the hassle of software maintenance). The problem I'm trying to solve at the moment is getting caller ID info passed over to the SIP phones when calls are placed. The caller ID is coming through as 'asterisk' which I assume is the default if nothing is present. So does anyone have any idea on how to get caller ID info passed over from the incoming calls on the FXO lines? Or even how to find out if the caller ID info is being sent by the Samsung? Thanks for any help. Ronny -- Ronny Adsetts Technical Director Amazing Internet Ltd, London t: +44 20 8607 9535 f: +44 20 8607 9536 w: www.amazinginternet.com Registered office: UK House, 82 Heath Road, Twickenham TW1 4BW Registered in England. Company No. 4042957 signature.asc Description: OpenPGP digital signature -- _ -- 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] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Sure thing! Bug #18302 has been opened (https://issues.asterisk.org/view.php?id=18302). Brett Woollum br...@woollum.com - Original Message - From: Sherwood McGowan sherwood.mcgo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 12:20:23 PM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables Sounds good mate, keep me posted, and let me know the issue number so I can check in on it :D Who knows, I might be able to offer some testing or somethin' for the digium guys or something -- _ -- 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] CallerID from Samsung PBX line on FXO
Ronny Adsetts wrote: Hi, I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX via an 8SLI analogue extension card in the Samsung and an Openvox A400P04 4-FXO card in the Asterisk box. It all works in that I can place calls in both directions from the office Samsung extensions and Asterisk SIP extensions. The only tricky bit was getting the FXO to detect hang-up from the Samsung correctly - a tweak on the Samsung config to make it hang up properly fixed this. So far Asterisk has been a joy to configure (apart from the default config which IMO should be stripped back to bare-bones and the config shipped as examples in the doc folder - this may be the fault of my distribution who's packages I used rather than get in to the hassle of software maintenance). The problem I'm trying to solve at the moment is getting caller ID info passed over to the SIP phones when calls are placed. The caller ID is coming through as 'asterisk' which I assume is the default if nothing is present. So does anyone have any idea on how to get caller ID info passed over from the incoming calls on the FXO lines? Or even how to find out if the caller ID info is being sent by the Samsung? Thanks for any help. Ronny Assuming the Samsung is using the standard US method, a butt set across the output of one of the lines in monitor mode should allow you to hear the FSK tones between the first and second ring. Or a cheap CallerID box should display as well. Also assume you have the Asterisk end set to wait for the signal. there should be a one ring delay in the response from Asterisk as it waits for the FSK signal. John Novack -- Dog is my Co-pilot -- _ -- 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] CallerID from Samsung PBX line on FXO
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
caller ID is coming through as 'asterisk' which I assume is the
default if nothing is present. So does anyone have any idea on how
to get caller ID info passed over from the incoming calls on the
FXO lines? Or even how to find out if the caller ID info is being
sent by the Samsung?
Assuming the Samsung is using the standard US method, a butt set
across the output of one of the lines in monitor mode should allow
you to hear the FSK tones between the first and second ring. Or a
cheap CallerID box should display as well.
Also assume you have the Asterisk end set to wait for the signal.
there should be a one ring delay in the response from Asterisk as it
waits for the FSK signal.
I've tried a variety of options but currently have the following in the
[channels] section of zapata.conf:
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=ring
sendcalleridafter=2
hidecallerid=no
I added the following to one of the FXO contexts:
exten = s,1,Verbose(1,Samsung 209 ${CALLERID(all)})
And I get the following:
Executing [...@samsung-209:1] Verbose(Zap/1-1, 1|Samsung 209 ) in new
stack
Not sure what means but I assume it's something like a null string?
Anyway, that's as successful as I've been, ie., not. :-).
Ronny
--
Ronny Adsetts
Technical Director
Amazing Internet Ltd, London
t: +44 20 8607 9535
f: +44 20 8607 9536
w: www.amazinginternet.com
Registered office: UK House, 82 Heath Road, Twickenham TW1 4BW
Registered in England. Company No. 4042957
signature.asc
Description: OpenPGP digital signature
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[asterisk-users] eSXI and Asterisk?
Hi Everyone, I don't have much experience with eSXI. I can really use some advise on how to run it without any trouble with Asterisk on CentOS VMs. First of all, is it a good option to run multiple hosted Asterisk instances on a VMware eSXI? or would you rather prefer something like Xen, proxmox, opennode, etc? (All SIP trunking, no PRI or Analogue) If there are limitations such as timing, reliability, I/O access, bad voice quality due to sharing resources please let me know. If you have experience working with it in production with Asterisk please let me know what type of fine tuning you do to get this running. 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] asterisk 1.8 fax woes
On 11/13/2010 4:36 AM, Jeremy Kister wrote: When a caller connects, asterisk switches to the fax context and hangs up the call. I was wrong, asterisk does not even switch to the fax extension- i added a noop, and it's not making it: exten = fax,1,NoOp( in fax extension ) exten = fax,n,Goto(fax,rx,1) the call ends before the noop. -- 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] CallerID from Samsung PBX line on FXO
At 05:56 AM 11/13/2010, you wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
exten = s,1,Verbose(1,Samsung 209 ${CALLERID(all)})
And I get the following:
Executing [...@samsung-209:1] Verbose(Zap/1-1, 1|Samsung 209
) in new stack
Try changing that line to:
exten = s,1,wait(1) or maybe wait(2)
That's what I had to do when I first set up Asterisk. Gives Asterisk
time to get the CID.
Ira
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[asterisk-users] Nat Issue - I think
Hi, I'm using qualify= on my asterisk server that provides outgoing pstn calls to a few companies. I've got one client in particular that has their own asterisk server which is connected to my server. This client seems to be having a nat issue. It's not a connectivity issue as i've tried constant pings and the line is up constantly. I'm getting the following: [2010-11-13 17:56:27] NOTICE[26082] chan_sip.c: Peer 'client_201' is now UNREACHABLE! Last qualify: 29 [2010-11-13 17:56:37] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (26ms / 5000ms) [2010-11-13 17:58:12] NOTICE[26082] chan_sip.c: Peer 'client _201' is now UNREACHABLE! Last qualify: 28 [2010-11-13 17:58:22] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (27ms / 5000ms) [2010-11-13 17:59:58] NOTICE[26082] chan_sip.c: Peer 'client _201' is now UNREACHABLE! Last qualify: 27 [2010-11-13 18:00:08] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (27ms / 5000ms) [2010-11-13 18:01:43] NOTICE[26082] chan_sip.c: Peer 'client _201' is now UNREACHABLE! Last qualify: 29 [2010-11-13 18:01:53] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (28ms / 5000ms) Looking at the sip log, on the client's server, the sip OPTIONS packets arent being received. Then suddenly the sip packets start being received again. (without sending out a new packet to open up the nat mapping). I've tried replacing the router because I thought it was faulty. Here is the SIP log from my server: http://pastebin.com/ZbDYGG9R Finally, I tried mapping port 5060 so to avoid NAT issues, but that didnt help. Could there be an ISP problem? Any assistance would be appreciated. Many thanks Dan -- _ -- 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 registering to ekiga.net
Hi! I want my PBX to be reachable at my ekiga.net account. It seems I am registered: vajna2*CLI sip show registry HostUsername Refresh StateReg.Time ekiga.net:5060 magwas 585 Registered Sat, 13 Nov 2010 13:48:22 However when others try to call mag...@ekiga.net, they find me unavailable. My asterisk is available when called directly. (As an aside note, the http://www.ekiga.net/status/presence.php?user=magwas shows me as available even when I am not registering.) What could be the problem? my sip.conf: [general] context=default ; Default context for incoming calls srvlookup=yes ;videosupport=yes allowoverlap=no ; Disable overlap dialing support. (Default is yes) realm=patyicivil.local ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) externip=188.36.152.83 srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all ; First disallow all codecs ;allow=g726 ;allow=g729 allow=speex allow=ulaw allow=alaw ; Allow codecs in order of allow=ilbc ; preference allow=gsm ;allow=h261 localnet=10.0.0.0/255.0.0.0 register = magwas:mypassw...@ekiga.net registertimeout=20 ; retry registration calls every 20 seconds (default) registerattempts=0 -- _ -- 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 1.8 fax woes
This does sound like something that should stay on Asterisk-users. On Sat, Nov 13, 2010 at 3:36 AM, Jeremy Kister asterisk...@jeremykister.com wrote: I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. So you made sure to remove the res_fax.so module that was there from 1.6.2? Tried cleaning out the modules directory then installing just the 1.8 modules to be safe? All my custom modules (including swift thanks darren!) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip set debug peer vgw1 (vgw1 is my cisco 1760 ata) http://jeremy.kister.net/tmp/fax/console.txt http://jeremy.kister.net/tmp/fax/messages.txt http://jeremy.kister.net/tmp/fax/sip.txt I've tried using the packaged app_fax_spandsp and also Digium's app_fax_digum for 1.8.0-rc1 -- no difference in behavior. Anyone have any ideas how I can get this fixed? Have you tried doing tests where you send all calls straight into ReceiveFax and disable faxdetect? That may help track down where the problem is at least. You can put a noop before the call to receivefax if you'd like, but keep it simple and don't do anything else for this part of the test. If you've got a paid for Fax license (as opposed to Free Fax) then you can also contact Digium Support. -- _ -- 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-stat v.2 and mysql 5.1.51
On 11/12/10 23:14, Joseph wrote: After upgrade to mysql 5.1.51 my asterisk-stat-v2 is not displaying correctly. Does anybody have a similar problem? Is it due to mysql-5.1.51 or the problem is with new glibc-2.11.2 ? -- Joseph I see that asterisk-stat-v2 was replaced with CDR-Stats The installation does not appear as easy as the old one. Does anybody have detailed instructions how to install it and integrate with asterisk? The installation documentation enclosed with the package is not of much help and most of it centered around one or two distros. -- Joseph -- _ -- 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] EXTENDED: Scheduled maintenance for various Asterisk community services
Between 10:00AM and 10:00PM CST on Saturday, November 13, the services below will experience extended outages as the servers that host them are upgraded and reconfigured: downloads.digium.com downloads.asterisk.org bamboo.asterisk.org packages.asterisk.org svn.digium.com svn.asterisk.org issues.asterisk.org reviewboard.asterisk.org wiki.asterisk.org code.asterisk.org We apologize for any inconvenience this may cause. The maintenance was originally expected to be completed by 1:00PM, but unforeseen complications have made it take longer than planned. -- _ -- 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] CallerID from Samsung PBX line on FXO
Ronny Adsetts wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
caller ID is coming through as 'asterisk' which I assume is the
default if nothing is present. So does anyone have any idea on how
to get caller ID info passed over from the incoming calls on the
FXO lines? Or even how to find out if the caller ID info is being
sent by the Samsung?
Assuming the Samsung is using the standard US method, a butt set
across the output of one of the lines in monitor mode should allow
you to hear the FSK tones between the first and second ring. Or a
cheap CallerID box should display as well.
Also assume you have the Asterisk end set to wait for the signal.
there should be a one ring delay in the response from Asterisk as it
waits for the FSK signal.
I've tried a variety of options but currently have the following in the
[channels] section of zapata.conf:
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=ring
sendcalleridafter=2
hidecallerid=no
I added the following to one of the FXO contexts:
exten = s,1,Verbose(1,Samsung 209 ${CALLERID(all)})
And I get the following:
Executing [...@samsung-209:1] Verbose(Zap/1-1, 1|Samsung 209 ) in new
stack
Not sure what means but I assume it's something like a null string?
Anyway, that's as successful as I've been, ie., not. :-).
Ronny
What method does the Samsung use to send CLID?
For US systems, the FSK signal is between the first and second ring, is
NOT dtmf, and cidstart=ring isn't correct
You need to state where in the world the samsung thinks it is, and how
it is configured to send CLID.
If it isn't US, then others will have to help
In the US Asterisk will insert the proper delay when you state
usecallerid=yes
John Novack
--
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[asterisk-users] dial plan and sip
Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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] dial plan and sip
Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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] A few questions regarding Asterisk 1.8.0
Hello, I have a few questions regarding Asterisk 1.8.0. If you can answer a question, please do so. Is Asterisk 1.8.0 stable enough for production environments? Is it possible (and if yes what is the best option) to use CDR MySQL with Asterisk 1.8.0? With 1.6.x we use the add-on package for that, however we could do something with scripts to do it (but I don't like the idea). If it is stable and there is a good option for CDR with MySQL we will start using it very soon. Regards, Mark -- _ -- 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] dial plan and sip
Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A few questions regarding Asterisk 1.8.0
From: Mark Scholten m...@streamservice.nl Hello, I have a few questions regarding Asterisk 1.8.0. If you can answer a question, please do so. Is Asterisk 1.8.0 stable enough for production environments? It appars to be so far we are testing and hoping to go production before the end of the year. Is it possible (and if yes what is the best option) to use CDR MySQL with Asterisk 1.8.0? With 1.6.x we use the add-on package for that, however we could do something with scripts to do it (but I don't like the idea). You can use the same MySQL method you are use to but if you want to use the new more extensive CEL method you will likely need to use ODBC to write to MySQL for now. You will also need to parse the new CEL format for the info you need. It is looking realy cool but it is taking a bit of work to intagrate it into our system. We will go live using the old CDR to MySQL for now. Please not that the addons are part of the main package now use menuselect to choose which ones you want to build. If it is stable and there is a good option for CDR with MySQL we will startusing it very soon. Good luck as with any new version there may be some bugs so if you bump up against ones report them so they can be fixed. Also don't just drop it into production with out testing it on a box for a bit. 1.8 has a lot of changes. Most appear to be for the better. Regards, Mark Regards Bryant -- _ -- 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] dial plan and sip
What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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] upgrade
i am running 1.4.37 and am hosted on Rackspace. I feel like a took a step back by using the Cloud server service since I am having a little trouble proving that my basic configuration is working. Nevertheless, I want to upgrade to 1.8. I use Centos 5.5 Anyone know of a good link that can help please? I searched Google and got confused by the options. Upgrade to 1.8. How please? -- _ -- 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] dial plan and sip
How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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] dial plan and sip
You get into asterisk by saying asterisk -r. You then up the verbosity by saying core set verbose 3 or some such number. You the call your number and you should see the steps of your dialplan execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
Jim, Thanks. But, no joy. I set to 3, then 5. I don't think I am getting registered somewhere. The console shows nothing. The call to the DID drops after 5 seconds or so. It does not ring. I know. Basic stuff. I really think the version of this code is not robust enough. My sip.conf and extensions.conf is very simple. On Sat, Nov 13, 2010 at 10:13 PM, Jim Dickenson dicken...@cfmc.com wrote: You get into asterisk by saying asterisk -r. You then up the verbosity by saying core set verbose 3 or some such number. You the call your number and you should see the steps of your dialplan execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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 -- _ -- 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] upgrade
On Sat, Nov 13, 2010 at 8:00 PM, Thomas Perron thomas.per...@gmail.comwrote: i am running 1.4.37 and am hosted on Rackspace. I feel like a took a step back by using the Cloud server service since I am having a little trouble proving that my basic configuration is working. Nevertheless, I want to upgrade to 1.8. I use Centos 5.5 Anyone know of a good link that can help please? I searched Google and got confused by the options. Upgrade to 1.8. How please? -- _ -- 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 When asking questions here, you should try and provide some details that round out your setup. So its in a cloud, are you using just SIP or other stuff as well? Any AGI's? Why can't you prove that your basic configuration is working? Have you read the blurb about pipes vs commas in extensions.conf in regards to compatibilty with 1.2, 1.4 and 1.6? if not read the guide that explains differences between 1.4 and 1.6. Confused by what options? -- _ -- 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] changing sip port
11 nov 2010 kl. 23.25 skrev Baha @ SH: Hello How can I run the sip service on asterisk on another port beside 5080? I mean asterisk will still take sip requests on port:5080 and another custom port, lets say port:6080 For UDP, we only have one port. You have to select. /O -- _ -- 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] OT: certificate for softphone
10 nov 2010 kl. 21.48 skrev Hans Witvliet: On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote: 6 nov 2010 kl. 15.30 skrev Hans Witvliet: Hi all, As stated in the subject, slightly off-topic, as it is not directly a Asterisk issue, but more SIP in general Because security in general, and specifically identification becomes more and more a subject for more concern, and Asterisk is capable of doing sip/TLS, i was wondering what more could be done to improve security. Specially softphones, might it be possible to employ etokens or smartcards for holding the certificates needed by TLS? Done before? In the SIP protocol there is support for TLS client certificates, much like in HTTP. Asterisk doesn't support it. You need to put a SIP proxy like Kamailio in front of Asterisk to get this kind of strong authentication. /O Am i that mistaken? I got the impression** that sip-registration of a phone could be done in the same way as client-authentication on apache: On the server-side you got the certificate holding your public key which is signed by a trusted third party (the CA), while you hold your private key on a smartcard or token. If you start your browser you are prompted for your pin-code. I was just hoping that there would be a softphone that could work the same way, two-factor authentication. I haven't seen any soft clients implementing this. Bria/Eyebeam may have it, but they've removed all TLS options from the GUI. As I said, the SIP protocol supports it. Kamailio supports it on the server side. Now we need clients that supports it. Now we're talking about authentication. For identity assurance, there's another set of standards called SIP Identity where you use TLS to sign your identity. The TLS is just between the phone and the first server. Identity is supposed to be something that follows the call to the callee. /O -- _ -- 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
