[asterisk-users] How to get ringing sound in outbound call in asterisk
I have two server
Server_A(outbound call) for agent login and agent make a outbound call from
here and pass into server Server_B call
extension.conf
exten = _91XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR)
exten = _91XX.,n,hangup()
Server_B[192.168.53.197] for call forwarding
extension.conf
exten = _911X.,1,ChanisAvail(${TRUNK_GRP3})
exten = _911X.,2,gotoif($[${AVAILCHAN} = ]?lbl_busy:)
exten =
_911X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten = _911X.,n,Gotoif($[${RECORDING_ENABLED}=Y]?lbl_dbc:lbl_dial)
exten =
_911X.,n(lbl_dbc),Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten = _911X.,n,Set(CALLERID(num)=${IDGCLI})
exten = _911X.,n,Set(FILENAME=${IDGTERMINAL}_${EXTEN:1}_${CALLTIME}.WAV)
exten = _911X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
exten = _911X.,n,Gotoif($[${IDGCALL}=]?lbl_setcall:lbl_sendevent)
exten = _911X.,n(lbl_setcall),Set(IDGCALL=0)
exten = _911X.,n(lbl_sendevent),Gotoif($[${DBTYPE}=SQL]?lbl_sql:)
exten = _911X.,n,Gotoif($[${DBTYPE}=MYSQL]?lbl_mysql:lbl_record)
exten = _911X.,n(lbl_sql),UserEvent(${CHANNEL}$DBEXEC$EXEC
udsp_vlog_start_record
'${CHANNEL}'#'${IDGTERMINAL}'#'${EXTEN:1}'#'${FILENAME}'#'${UNIQUEID}'#'${CALLERID(num)}'#'O'#${VLOGSERVER}#'${HTTPPATH}${RECORDFILENAME}'#${IDGCALL}$)
exten = _911X.,n,Goto(lbl_record)
exten = _911X.,n(lbl_mysql),UserEvent(${CHANNEL}$DBEXEC$CALL
udsp_vlog_start_record
('${CHANNEL}'#'${IDGTERMINAL}'#'${EXTEN:1}'#'${FILENAME}'#'${UNIQUEID}'#'${CALLERID(num)}'#'O'#${VLOGSERVER}#'${HTTPPATH}${RECORDFILENAME}'#${IDGCALL})$)
exten =
_911X.,n(lbl_record),MixMonitor(${RECORDING_PATH_OUT_SREI}${RECORDFILENAME})
exten = _911X.,n(lbl_dial),Set(ChanLength=${LEN(${AVAILCHAN})})
exten = _911X.,n,Set(NewChannel=${AVAILCHAN:0:$[${ChanLength}-2]})
exten = _911X.,n,Dial(${NewChannel}/${EXTEN:3},,tToR)
exten = _911X.,n,hangup()
exten = _911X.,n(lbl_busy),Busy()
I'm not able listened ringing sound when i make outbound call.
Regards
Akhilesh
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Re: [asterisk-users] Asterisk API
On 10 January 2014 17:12, James Wystead szilvertho...@gmail.com wrote: Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen You could use asterisk realtime architecture and use your favourite database to hold peer/voicemail/dialplan configuration. https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 API
Good Day, Ishfaq; This may be a much better idea than the REST API. Correct me if I'm wrong, but the concept is this: You write to the database, and this gives the same result as perhaps modifying the dialplan, sip, voicemail, etc *without* having to physically modify the extensions.conf, sip.conf, voicemail.conf? Am I on the right track? Thanks! On Mon, Jan 13, 2014 at 4:16 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 10 January 2014 17:12, James Wystead szilvertho...@gmail.com wrote: Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen You could use asterisk realtime architecture and use your favourite database to hold peer/voicemail/dialplan configuration. https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API
Yes, this would most likely be a better solution (REST API is in Asterisk 12), just be careful about putting your dialplan in a Real Time Database. You sometimes have to do things a little different if your dialplan is in the RTDB. As well you need to make sure security is locked down when using this approach, but security is needed for all approaches. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Wystead Sent: Monday, January 13, 2014 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk API Good Day, Ishfaq; This may be a much better idea than the REST API. Correct me if I'm wrong, but the concept is this: You write to the database, and this gives the same result as perhaps modifying the dialplan, sip, voicemail, etc *without* having to physically modify the extensions.conf, sip.conf, voicemail.conf? Am I on the right track? Thanks! On Mon, Jan 13, 2014 at 4:16 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 10 January 2014 17:12, James Wystead szilvertho...@gmail.com wrote: Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen You could use asterisk realtime architecture and use your favourite database to hold peer/voicemail/dialplan configuration. https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configurati on -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 tel:%2B44%20%280%29845%20004%204994 f: +44 (0)161 660 9825 tel:%2B44%20%280%29161%20660%209825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On 01/10/2014 08:33 PM, gm1 wrote: On 01/10/2014 04:01 PM, Matthew Jordan wrote: On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote: On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:portnumber then not until about 6 seconds later I see this 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:diffportnumber and immediately hear audio what appears to be an issue is that the RTP link(audio) setup is delayed. Anyone have suggestions on how to fix this issue? If the RTP source address/port is changing, then Asterisk is receiving RTP packets from two different sources and is waiting for one of them to stabilize before it picks the actual source of the media stream. That's by design, as the locking in of the RTP source prevents a media injection attack. You can tweak how Asterisk does this using two settings in rtp.conf: ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is ; enabled by default. ; strictrtp=yes ; Number of packets containing consecutive sequence values needed ; to change the RTP source socket address. This option only comes ; into play while using strictrtp=yes. Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. ; probation=8 Matt Matt, What if any risk do i have with setting strictrtp=no with nat=no on a local network i.e.: 192.168.1.x ? pc I changed strictrtp=no and restarted asterisk, no difference in delayed audio ... still near 6 seconds. In cli when I answer the incoming call I see asterisk immediately show answer. Perhaps this issue is caused by something other than the strictrtp setting? what are all the possible settings for strictrtp=??? -- _ -- 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] How to read IRQs and timing slips values
2014/1/10 Shaun Ruffell sruff...@digium.com On Fri, Jan 10, 2014 at 08:34:57PM +0100, Olivier wrote: 2014/1/10 Shaun Ruffell sruff...@digium.com You've configured the card to recover timing from the provider? I'm not sure but I don't think so as I've just configured the card with: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 echocanceller=oslec,32-46,48-62 Span1 is the one direct to provider equipement. Span2 is thh one that was connected to HiPath and which is simply unplugged That looks correct. You might want to check your cables next. Do you only get timing slips when connected to the provider, or to the HiPath as well? For a couple of days now, I'm only connected to the provider. Just in case, I commented out in system.conf and dahdi-channels.conf, any reference to the HiPath span and ran a dahdi restart now: unfortunately, I'm still observing timing slips at a rate of 1220 slips/hour. I have a spare TE4xx board. Shall I replace current TE2xx with this spare and see if things improve or would you advise an other check ? Regards -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] How to read IRQs and timing slips values
On Mon, Jan 13, 2014 at 04:13:49PM +0100, Olivier wrote: 2014/1/10 Shaun Ruffell sruff...@digium.com On Fri, Jan 10, 2014 at 08:34:57PM +0100, Olivier wrote: 2014/1/10 Shaun Ruffell sruff...@digium.com You've configured the card to recover timing from the provider? I'm not sure but I don't think so as I've just configured the card with: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 echocanceller=oslec,32-46,48-62 Span1 is the one direct to provider equipement. Span2 is thh one that was connected to HiPath and which is simply unplugged That looks correct. You might want to check your cables next. Do you only get timing slips when connected to the provider, or to the HiPath as well? For a couple of days now, I'm only connected to the provider. Just in case, I commented out in system.conf and dahdi-channels.conf, any reference to the HiPath span and ran a dahdi restart now: unfortunately, I'm still observing timing slips at a rate of 1220 slips/hour. I have a spare TE4xx board. Shall I replace current TE2xx with this spare and see if things improve or would you advise an other check ? If you have another board, yes, you could try. But I would recommend checking all your cables, etc. Also, while highly unlikely, I've heard of cases in the past where some smaller providers were expecting to source timing from customer premise PBX (since they were acting as a SIP gateway on the backend). Your provider does expect you to source their clock? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to read IRQs and timing slips values
On 01/13/2014 11:39 AM, Shaun Ruffell wrote: If you have another board, yes, you could try. But I would recommend checking all your cables, etc. Also, while highly unlikely, I've heard of cases in the past where some smaller providers were expecting to source timing from customer premise PBX (since they were acting as a SIP gateway on the backend). Check the T1 cable doesn't pass any high EMI area's like a power supply. Adrian -- _ -- 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] How to read IRQs and timing slips values
Ask your carrier to test the circuit. Often HDLC errors, especially with modern cards, are caused by a dirty T-1 not a PBX or card issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Serafini Sent: Monday, January 13, 2014 1:19 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to read IRQs and timing slips values On 01/13/2014 11:39 AM, Shaun Ruffell wrote: If you have another board, yes, you could try. But I would recommend checking all your cables, etc. Also, while highly unlikely, I've heard of cases in the past where some smaller providers were expecting to source timing from customer premise PBX (since they were acting as a SIP gateway on the backend). Check the T1 cable doesn't pass any high EMI area's like a power supply. Adrian -- _ -- 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] help with Cepstral 6 and Asterisk 11
Another option is to use an MRCP server like UniMRCP along with the Cepstral plugin. One very nice thing about this approach is that there is less 'cepstral version' - 'asterisk version' dependency, which is a problem with the current app_swift module (each app_swift version is designed to work on specifically one version of asterisk and one version of cepstral). Cepstral provides details here: http://www.cepstral.com/en/telephony/mrcp Information on the open-source uniMRCP can be found here: http://www.unimrcp.org/ Information for connecting asterisk to uniMRCP can be found here (although it seems to be having issues ATM): http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP -Justin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg Sent: Friday, January 10, 2014 1:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11 Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take more than a couple of minutes to compile, install, and configure the package. Maybe some things need to get adjusted as the author has abandoned the project (at least there is no longer a project web page) and the latest sources are about 2 years old. Building from sources is not that difficult, but if you don't have a proper configure script you are responsible that all prerequisites are met, which can be time consuming if you don't know your distro well enough. Here, there is no configure script and some things, which might be invalid for your machine, are hand coded inside the Makefile. Nothing spectacular, but you could end up asking a lot more questions that have nothing to do with asterisk. jg -- _ -- 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] From: Unavailable sip:aster...@server.com; tag=as120a1079.
Hello Everyone, Calls that are private name private number have the following TO header: From: Unavailable sip:aster...@server.com;tag=as120a1079. Don't tell anyone, but we are trying to put on a We're big enough to own the pricey softswitch look. Even though I would pick a OpenSIPS + Asterisk combo over a Metaswitch any day. Three words Service Licence Agreement. Anyhow long story short, is there any way to change the asterisk part only for the calls that are private? Everything else can stay the same.. Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] From: Unavailable sip:aster...@server.com; tag=as120a1079.
Correction, and by TO, I mean FROM header :) -- _ -- 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] Does cdr adaptive odbc re-connect automatically after a long idle time?
On Sat, Jan 11, 2014 at 4:56 PM, Charles Wang lazy.char...@gmail.com wrote: Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR messages in the CLI at this period. Could you please tell me which settings are wrong? Why dose my odbc connection not re-connect to MySQL automatically? I checked the setting below: CLI: ubuntu*CLI cdr show status Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log unanswered calls: Yes Log congestion: Yes * Registered Backends --- cdr-custom Adaptive ODBC csv ubuntu*CLI odbc show all ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2014-01-11 18:16:40 Pooled: Yes Limit: 1000 Connections in use: 0 -- /etc/asterisk/cdr.conf lists below: [general] enable=yes unanswered = yes congestion = yes endbeforehexten=yes [csv] usegmtime=no; log date/time in GMT. Default is no loguniqueid=yes ; log uniqueid. Default is no loguserfield=yes ; log user field. Default is no accountlogs=yes ; create separate log file for each account code. Default is yes -- /etc/odbc.ini [asterisk-connector] Description = MySQL connection to 'asterisk' database Driver= MySQL Database = mydatabase Server= localhost UserName = root Password = mypassword Port = 3306 Socket= /var/run/mysqld/mysqld.sock -- /etc/asterisk/res_odbc.conf lists below: [ENV] [asterisk] enabled = yes dsn = asterisk-connector password = mypassword pre-connect = yes sanitysql = select 1 pooling = yes idlecheck = 30 share_connections = yes limit = 1000 connect_timeout = 60 negative_connection_cache = 600 -- /etc/asterisk/cdr_adaptive_odbc.conf lists below: [cdr] connection=asterisk table=cdr alias start = calldate alias phoneno = phoneno alias userid = userid alias callerid = callerid I would be inclined to check the database side over asterisk. We use almost the same setup and don't have any issues. We go some time 12 hours between calls. Once thing you could do is enable debug logs and see what Asterisk is doing when the odbc connection is down. EG: it should be attempting to reconnect. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Well, if you want anybody to call you, you need to leave it open to the public. Meaning, you can't really secure it. Obviously, don't have any outbound trunks configured on the box so that the only location some could dial would be your extension. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] How to read IRQs and timing slips values
On Thu, Jan 9, 2014 at 12:01 PM, Olivier oza.4...@gmail.com wrote: Hi, On a Asterisk 1.8.12 system working OK for months (100k calls proceed), users are complaining for bad audio. My setup is: PSTN --E1/PRI --- Asterisk --- E1/PRI--- Siemens HiPath ---E1/PRI --- PSTN asterisk -rx dahdi show version DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx pri show version libpri version: 1.4.12 A quick glance at Asterisk logs shows lines like this: [2014-01-09 17:19:34] NOTICE[26034]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 [2014-01-09 17:19:35] NOTICE[26035]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 2 [2014-01-09 17:19:49] NOTICE[26035]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 2 I read an old thread inviting an admin to check for shared IRQs and timing slips. My questions are: 1. cat /proc/interrupts 's output is: # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 0: 90147 0 0 0 0 0 0 0 IO-APIC-edge timer 1: 2 0 0 0 0 0 0 0 IO-APIC-edge i8042 8: 0 0 1 0 0 0 0 0 IO-APIC-edge rtc0 9: 0 0 0 0 0 0 0 0 IO-APIC-fasteoi acpi 12: 4 0 0 0 0 0 0 0 IO-APIC-edge i8042 14: 93 0 0 0 0 0 0 0 IO-APIC-edge ata_piix 15: 0 0 0 0 0 0 0 0 IO-APIC-edge ata_piix 16: 3378646209 3378695076 3378691115 3378697362 3378691116 3378706831 3378710635 3378702358 IO-APIC-fasteoi wct2xxp Can I positively conclude that my Dahdi PRI board IS NOT sharing IRQ (which is good) ? 2. What would you suggest reading the following output ? cat /proc/dahdi/2 Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) HDB3/CCS Timing slips: 175319 32 TE2/0/2/1 Clear (In use) (EC: VPMOCT064 - INACTIVE) 33 TE2/0/2/2 Clear (In use) (EC: VPMOCT064 - INACTIVE) 34 TE2/0/2/3 Clear (In use) (EC: VPMOCT064 - INACTIVE) 35 TE2/0/2/4 Clear (In use) (EC: VPMOCT064 - INACTIVE) 36 TE2/0/2/5 Clear (In use) (EC: VPMOCT064 - INACTIVE) 37 TE2/0/2/6 Clear (In use) (EC: VPMOCT064 - INACTIVE) 38 TE2/0/2/7 Clear (In use) (EC: VPMOCT064 - INACTIVE) 39 TE2/0/2/8 Clear (In use) (EC: VPMOCT064 - INACTIVE) 40 TE2/0/2/9 Clear (In use) (EC: VPMOCT064 - INACTIVE) 41 TE2/0/2/10 Clear (In use) (EC: VPMOCT064 - INACTIVE) 42 TE2/0/2/11 Clear (In use) (EC: VPMOCT064 - INACTIVE) 43 TE2/0/2/12 Clear (In use) (EC: VPMOCT064 - INACTIVE) 44 TE2/0/2/13 Clear (In use) (EC: VPMOCT064 - INACTIVE) 45 TE2/0/2/14 Clear (In use) (EC: VPMOCT064 - INACTIVE) 46 TE2/0/2/15 Clear (In use) (EC: VPMOCT064 - INACTIVE) 47 TE2/0/2/16 HDLCFCS (In use) (EC: VPMOCT064 - INACTIVE) 48 TE2/0/2/17 Clear (In use) (EC: VPMOCT064 - INACTIVE) 49 TE2/0/2/18 Clear (In use) (EC: VPMOCT064 - INACTIVE) 50 TE2/0/2/19 Clear (In use) (EC: VPMOCT064 - INACTIVE) 51 TE2/0/2/20 Clear (In use) (EC: VPMOCT064 - INACTIVE) 52 TE2/0/2/21 Clear (In use) (EC: VPMOCT064 - INACTIVE) 53 TE2/0/2/22 Clear (In use) (EC: VPMOCT064 - INACTIVE) 54 TE2/0/2/23 Clear (In use) (EC: VPMOCT064 - INACTIVE) 55 TE2/0/2/24 Clear (In use) (EC: VPMOCT064 - INACTIVE) 56 TE2/0/2/25 Clear (In use) (EC: VPMOCT064 - INACTIVE) 57 TE2/0/2/26 Clear (In use) (EC: VPMOCT064 - INACTIVE) 58 TE2/0/2/27 Clear (In use) (EC: VPMOCT064 - INACTIVE) 59 TE2/0/2/28 Clear (In use) (EC: VPMOCT064 - INACTIVE) 60 TE2/0/2/29 Clear (In use) (EC: VPMOCT064 - INACTIVE) 61 TE2/0/2/30 Clear (In use) (EC: VPMOCT064 - INACTIVE) 62 TE2/0/2/31 Clear (In use) (EC: VPMOCT064 - INACTIVE) 3. As shown above, my box has two connections with PSTN (same provider for both): one direct, one through an HiPath PBX. How can I double check timing slips don't come from inconsistency between both clock sources ? My first thought would be to unplug the link between Asterisk and HiPath and compare two /pro/dahddi/1 outputs. Thoughts ? Regards I basically had the same issue as you for one of my sites. I tried everything under the sun to figure it out, change cables, loop back test, change out hardware, clocking, etc. In the end I had to upgrade dahdi to 2.7+ and the issue went away. Never did figure out the real problem, but to this day I think the issue was a delay on the frames from the PCI bus
Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On 14-01-14 02:36, Paul Belanger wrote: On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Well, if you want anybody to call you, you need to leave it open to the public. Meaning, you can't really secure it. Obviously, don't have any outbound trunks configured on the box so that the only location some could dial would be your extension. Thanks for your feedback Paul. The not having outbound trunks is going to be a challenge. So next to fail2ban I guess I'll cook up some dialplan logic that records IP addresses, keeps track of the amount of failed password attempts etc. and block the offending IP addresses together with max simultaneous outband calls and anything else I can think of to beef up security and limit potential damage. Thanks, Patrick -- _ -- 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
