Re: [asterisk-users] Asterisk NAT friendly settings
On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net wrote: I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia Have you added directmedia=no? Ish -- 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] Cisco2811 - E1 Pri - Asterisk (solved)
Le 18/12/2013 11:51, Administrator TOOTAI a écrit : Hi all, I face a strange problem. I'm in France using Completel as operator for the E1 line. I move a client from ccm to Asterisk keeping the 2811 gateway. Set up is complete, outgoing and incoming calls are sended to the 2811. The problem is that in 90% of the time I get a 500server error back from 2811 when placing a call. Incoming calls are OK. The 2811 was configured in primary-4ess (?) for ccm. I changed it to primary-net5. The ios is c2800nm-spservicesk9-mz.123-11.T5.bin If someone had any hint, would be great to share. For the archives: problem was solved by adding a prefix on outgoing call to Cisco (eg 9) and adding in dial-peer voice xxx pots destination-pattern 9T -- Daniel -- _ -- 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] Caller ID not real nor showing in call logs.
I have a multi tenant asterisk box where on tenant is receiving calls from the caller ID as1as and they cannot pickup the call. The caller ID also does not show up in the call log. Thoughts? Thanks, --Eric -- _ -- 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] Call duration limit ? Calls end after 15 minutes...
Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Using Asterisk 1.8.12.2 Kind regards, 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
Re: [asterisk-users] Call duration limit ? Calls end after 15 minutes...
Try setting canreinvite yes on that trunk it worked on trunks I had Some providers send a reinvite after 15 min and if Asterisk doesn't respond then it disconnects the call something like that -Original Message- From: Jonas Kellens jonas.kell...@telenet.be Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 08 Jan 2014 16:07:22 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Call duration limit ? Calls end after 15 minutes... -- _ -- 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 NAT friendly settings
On 1/8/2014 4:17 AM, Ishfaq Malik wrote: On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia Have you added directmedia=no? Nope, I'll look into that. -- _ -- 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] Call duration limit ? Calls end after 15 minutes...
Am 08.01.2014 16:07, schrieb Jonas Kellens: Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Look at session-timers in sip.conf. I had to set it to refuse for a specific provider because they are a little incompetent. Drawback is that a call can show as going on forever if the BYE message is lost due to network problems. -- _ -- 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] Call duration limit ? Calls end after 15 minutes...
On 08-01-14 16:47, Markus wrote: Am 08.01.2014 16:07, schrieb Jonas Kellens: Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Look at session-timers in sip.conf. I had to set it to refuse for a specific provider because they are a little incompetent. Drawback is that a call can show as going on forever if the BYE message is lost due to network problems. Are SIP session timers also present in IP-phones ? Or is this only a setting in a SIP-server and not in a SIP client like an IP-phone ? 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] (CALL FILES to Local Channel)billsec Zero in cdr via cdr_adaptive_odbc
Hi, all Sorry for null subject last mail. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it Asterisk11. I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database mydatabase) via cdr_adaptive_odbc. The SIP/A221 is another asterisk machine named it Elastix24. I have two BIG QUESTIONs about cdr_adaptive_odbc. First, I have answered call from Elastix24 and I can listen the music file played from Asterisk11. In another word, this call should be answered and its billsec is greater than 0. Second, if I don't want to use forkcdr(), how to config it and I can get another cdr record that call from SIP/A221(Elastix24) to my Exten:77? I know that the outgoing file will make a call to Local Channel and try to Dial SIP/A221. If it answered, this old channel should be hangup and generate another new channel to connect to Extension:77(my callback exten). I can't find two cdr records in mycdr table. mysql select * from gvl_cdr; +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:37:01 | | |77 | from-internal-out-7 | Local/77@from-internal-out-7-;2 | SIP/A221- | Dial| SIP/A221/77,30 | 17 | 0 | ANSWERED| 3 | | | 1389163021.1 | 1389163021.0 | 1| | 77 | | 7 | Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the 3th one). mysql select * from gvl_cdr; +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel| dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:34:04 || | 77| from-internal-out-7 | Local/77@from-internal-out-7-;2| SIP/A221- | Dial| SIP/A221/77,30| 15 | 0 | ANSWERED|3 | | | 1389162844.1 | 1389162844.0 | 1| | 77 | | 7 | | 2014-01-08 14:34:04 | device 1000| 1000| 77| from-6 | Local/77@from-internal-out-7-;1| | ForkCDR | | 20 | 5 | ANSWERED|3 | | | 1389162844.0 | 1389162844.0 | 0| | 77 | | 7 | | 2014-01-08 14:34:24 | device 77 | 77 | 77| from-6 | Local/77@from-internal-out-7-;1| | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 | 0 | NO ANSWER |3 | | | 1389162844.0 | 1389162844.0 | 3| | | | 0 | - /var/spool/asterisk/outgoing/77.call Channel:Local/77@from-internal-out-7 WaitTime:30 Context:from-6 Extension:77 Priority:1 Set:CLID= Set:EXT=77 Set:USERID=7 --
Re: [asterisk-users] Caller ID not real nor showing in call logs.
logs ? full log containing the call? On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote: I have a multi tenant asterisk box where on tenant is receiving calls from the caller ID as1as and they cannot pickup the call. The caller ID also does not show up in the call log. Thoughts? Thanks, --Eric -- _ -- 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 1.6.2.x Keeping NAT Alive
Richard - Perfect, Thanks for the pointer!Regards,Fred Original Message Subject: Re: [asterisk-users] Asterisk 1.6.2.x Keeping NAT Alive From: Richard Mudgett rmudg...@digium.com Date: Tue, January 07, 2014 4:42 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On Tue, Jan 7, 2014 at 4:32 PM, fred.robin...@sipfusion.com wrote: I'm running Asterisk 1.6.2.10, and I'm having the issue described here; https://issues.asterisk.org/jira/browse/ASTERISK-15721The last note says Patch "heap-fix.rev2.diff" was uploaded - was this Patch released, I don't see a reference for it in the thread? Anyone download patch and install it? The issue you reference shows that the patch was committed to the v1.6.2 branch with SVN revision -r261498.The commit message mentions the patch file but it does not appear to be attached to the issue. You can use SVN to generate a diff file of the committed change:svn diff -c261498 http://svn.asterisk.org/svn/asterisk change.patch Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Caller ID not real nor showing in call logs.
Does not show up in the cdr log. I am going to enable an asterisk cli dump tonight and try to catch it. I am thinking its a straight sip attack or IP attach on the sip client vs a real call or problem with asterisk. It's also a polycom IP 335 Thanks, --Eric Sent from my phone. On Jan 8, 2014, at 12:02 PM, Tiago Geada tiago.ge...@gmail.com wrote: logs ? full log containing the call? On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote: I have a multi tenant asterisk box where on tenant is receiving calls from the caller ID as1as and they cannot pickup the call. The caller ID also does not show up in the call log. Thoughts? Thanks, --Eric -- _ -- 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] Setting CDR variables for all linked channels
not sure about dial, but I Set(__var=value); and in each piece of dialplan I set CDR(var=value); On 31 December 2013 00:00, Igor Katson igor.kat...@gmail.com wrote: Hi, when one does Set(CDR(var)=value) in dialplan, the value is only set for one record in the cdr table, but not the linked ones (the ones with the same linkedid). E.g. if you do something like same = n, Set(CDR(var)=value) same = n,Dial(Local/somethingLocal/something2) like only the original CDR record with have var set to value, but the ones created from Dial won't. Is it possible to set the CDR variables in all the linked channels? P.S. And is it possible to find out by the CDR logs, if the originating call is in progress? 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 -- _ -- 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] help with Cepstral 6 and Asterisk 11
Hello, I recently purchased the Cepstral 6 text-to-speech engine (swift), and am now wondering if I should have bought something else. I would like to use Cepstral text to speech like some people use the Festival() or Flite() applications. For example, when I do a core show application flite at the CLI, flite is available to me: localhost*CLI core show application flite -= Info about application 'Flite' =- [Synopsis] Say text to the user, using Flite TTS engine [Description] Flite(text[,intkeys]): This will invoke the Flite TTS engine, send a text string, get back the resulting waveform and play it to the user, allowing any given interrupt keys to immediately terminate and return the value, or 'any' to allow any number back. I would like to do the same thing with Cepstral, and am trying to use app_swift available from: https://github.com/darrensessions/app_swift However, I am not able to get app_swift to compile. I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I am wondering if anyone else out there has been able to get app_swift working with Asterisk 11 and could share any tricks they used to get it installed? Brandon -- _ -- 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] is this expected behaviour?
i noticed in asterisk 10.12.3, i get messages like this: [2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite: Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63 but not mentioning attacker ip (to be used for fail2ban) is this expected? -- _ -- 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] Setting CDR variables for all linked channels
Thanks, Tiago! By the way that's exactly the workaround I came to myself. On Wed, Jan 8, 2014 at 2:35 PM, Tiago Geada tiago.ge...@gmail.com wrote: not sure about dial, but I Set(__var=value); and in each piece of dialplan I set CDR(var=value); On 31 December 2013 00:00, Igor Katson igor.kat...@gmail.com wrote: Hi, when one does Set(CDR(var)=value) in dialplan, the value is only set for one record in the cdr table, but not the linked ones (the ones with the same linkedid). E.g. if you do something like same = n, Set(CDR(var)=value) same = n,Dial(Local/somethingLocal/something2) like only the original CDR record with have var set to value, but the ones created from Dial won't. Is it possible to set the CDR variables in all the linked channels? P.S. And is it possible to find out by the CDR logs, if the originating call is in progress? 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 -- _ -- 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] is this expected behaviour?
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Framework -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Al lists Sent: Wednesday, January 08, 2014 10:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] is this expected behaviour? i noticed in asterisk 10.12.3, i get messages like this: [2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite: Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63 but not mentioning attacker ip (to be used for fail2ban) is this expected? -- _ -- 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
