[asterisk-users] Duplicate incoming channel into two outgoing channels
Hi! I have strange requirement: a incoming call should be duplicated to two outgoing calls (to two voice recorders). On the incoming channel we only receive RTP, on the two outgoing channel we only send RTP. I thought of: incoming call - originate: make outgoing call to recorder 1 and put the channel into a confBridge - originate: make outgoing call to recorder 2 and put the channel into the same confBridge - put the incoming channel into the same confBridge (as Admin, terminate conference when admin leaves) To optimize the conference processing I would configure the outgoing channels to listen only. This should lower CPU usage. Of course I need also some extra logic to periodically check if the outgoing channels still exists and if not, recreate the outgoing channels. Is somebody aware of a nicer solution for the problem? Thanks Klaus -- _ -- 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] Changing to the linear strategy currently requires asterisk to be restarted
Hello, using asterisk 1.8.12.2 and realtime architecture with mysql. I get the following message on CLI when changing the value in the strategy /[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted.// //[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted./ Can this be done without restarting asterisk ? Is this also the case in higher Asterisk versions ? For example Asterisk 1.8.24 ? 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] Duplicate incoming channel into two outgoing channels
What do you mean with voice recorders? Voice mail, if nobody answers, or do want to monitor calls? 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
Re: [asterisk-users] Spammer direct replying to those posting on the users list
[Intentionally ignoring the Reply-to header in this reply. And yes, this is on-list] On Tue, Mar 25, 2014 at 03:15:22PM +, A J Stiles wrote: On Tuesday 25 Mar 2014, Digium's Asterisk Development Team wrote: We apparently have a spam bot subscribed to the list and replying *directly* to anyone who posts on the list. The e-mail address I use for this mailing list is asterisk_l...@earthshod.co.uk ; so I used the following procmail recipe. This filters out anything being sent to that address *without* a Received: header mentioning lists.digium.com: :0 * ^To.*asterisk_list * !^Received.*lists.digium.com asterisk_unwanted What if I wanted to reply to one of your messages off-list? My message would end up in asterisk_unwanted. (when I am satisfied that it does not lose anything legitimate, I probably will change the last line to /dev/null .) Or even worse. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spammer direct replying to those posting on the users list
On Wednesday 26 Mar 2014, Tzafrir Cohen wrote: [Intentionally ignoring the Reply-to header in this reply. And yes, this is on-list] What if I wanted to reply to one of your messages off-list? My message would end up in asterisk_unwanted. And it did -- which is why I have added a special note in my .signature file (and why I didn't dump rejected messages straight into /dev/null/ from the beginning; the first rule of procmail seems to be, you probably missed something). A spammer probably isn't going to go to that sort of trouble. But if you can think of anything else I can pick up on that will improve the reliability of identifying legitimate off-list mail, I'm open to suggestions. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Numbers hackers call
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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] Numbers hackers call
972 is Israel See: http://en.wikipedia.org/wiki/List_of_country_calling_codes#Ordered_by_code -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Wednesday, March 26, 2014 11:05 AM To: Asterisk Users List Subject: [asterisk-users] Numbers hackers call I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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] Numbers hackers call
On 03/26/2014 10:05 AM, Michelle Dupuis wrote: I see a lot of attempts by hackers to call 00972595301123or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... I show that as Israel Cellular Jawall. j -- _ -- 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] Numbers hackers call
Hi The 11 bit is them thinking there's some prefix which will cause your PBX to become an open relay. The number (97259) is a Palestine Mobile number. These's a lot of hacking attempts coming from Palestine and this type of number probably has some revenue generation properties to it. Regards Ish On 26 March 2014 15:05, Michelle Dupuis mdup...@ocg.ca wrote: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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 -- 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] Numbers hackers call
http://en.wikipedia.org/wiki/Telephone_numbers_in_Israel Looks like it a mobile in Palestine - sure someone from Israel can tell us more 2014-03-26 16:05 GMT+01:00 Michelle Dupuis mdup...@ocg.ca: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Numbers hackers call
The number is not important i think. You need to block those country's you never use to connect to your asterisk system. I bet this call is made from palestine/israel too. Best regards. Emiliano Enviado desde mi BlackBerry de Personal (http://www.personal.com.ar/) -Original Message- From: Lenz Emilitri lenz.lo...@gmail.com Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 26 Mar 2014 16:14:02 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: Re: [asterisk-users] Numbers hackers call http://en.wikipedia.org/wiki/Telephone_numbers_in_Israel Looks like it a mobile in Palestine - sure someone from Israel can tell us more 2014-03-26 16:05 GMT+01:00 Michelle Dupuis mdup...@ocg.ca: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Verbose only one context
Hi It's possible in Asterisk 1.8 enable verbose only in one context or extension? thanks Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 -- _ -- 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] Secure audio cannot be provided
Hi Everyone. I am getting this error WARNING[31977][C-0009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer From the sdp can anyone suggest why secure audio cannot be provided v=0 o=- 6611325078116277019 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l m=audio 34054 UDP/TLS/RTP/SAVPF 111 103 104 0 8 107 106 105 13 126 c=IN IP4 10.1.1.2 a=rtcp:34054 IN IP4 10.1.1.2 a=candidate:53234734 1 udp 2113937151 10.1.1.2 34054 typ host generation 0 a=candidate:53234734 2 udp 2113937151 10.1.1.2 34054 typ host generation 0 a=candidate:4062413514 1 udp 2113937151 192.168.122.1 60784 typ host generation 0 a=candidate:4062413514 2 udp 2113937151 192.168.122.1 60784 typ host generation 0 a=candidate:1303359710 1 tcp 1509957375 10.1.1.2 0 typ host generation 0 a=candidate:1303359710 2 tcp 1509957375 10.1.1.2 0 typ host generation 0 a=candidate:3164634682 1 tcp 1509957375 192.168.122.1 0 typ host generation 0 a=candidate:3164634682 2 tcp 1509957375 192.168.122.1 0 typ host generation 0 a=ice-ufrag:iZyQ8Egkyi8hPKah a=ice-pwd:kQ91vFMHVr2lOkZfjGDLSfO+ a=ice-options:google-ice a=fingerprint:sha-256 81:DE:7E:6F:2B:88:8F:F3:30:82:92:DF:CB:FC:4B:63:BB:2E:BA:85:48:2B:B5:A6:C3:50:A1:42:E4:69:0E:91 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:haR/UikskQr/AIrry5udqINI1hYfc5zY2I6jrkKT a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:waQfKIHI9UyjPVI0vrcUREDbSVZdtfCtRQK71/Ks a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2121187131 cname:RXEEP3aaYIHOpxRX a=ssrc:2121187131 msid:YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4la0 a=ssrc:2121187131 mslabel:YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l a=ssrc:2121187131 label:YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4la0 - --- (14 headers 43 lines) --- Sending to 10.1.1.2:6443 (no NAT) Sending to 10.1.1.2:6443 (no NAT) Using INVITE request as basis request - 88vdo7l2kgd0eufrfd7q Found peer 'webrtc' for 'webrtc' from 10.1.1.2:42149 == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 111 Found RTP audio format 103 Found RTP audio format 104 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 107 Found RTP audio format 106 Found RTP audio format 105 Found RTP audio format 13 Found RTP audio format 126 Found audio description format opus for ID 111 Found unknown media description format ISAC for ID 103 Found unknown media description format ISAC for ID 104 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CN for ID 107 Found unknown media description format CN for ID 106 Found unknown media description format CN for ID 105 Found audio description format CN for ID 13 Found audio description format telephone-event for ID 126 [Mar 26 14:48:23] WARNING[31977][C-0009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer -- _ -- 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] Numbers hackers call
On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.ca wrote: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... It’s an international call to +972595XX, tried with the 00, 001 and no prefix What is confusing? Steve-- _ -- 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 12.1.1 - Having trouble setting up PJSIP
Thanks Joshua. Submitted issue ASTERISK-23539 with the information. I verified my pjproject is up to date and included the latest git log commit I have just in case. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, March 25, 2014 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP Dan Cropp wrote: I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down the problem. It certainly shouldn't crash no matter what you do. Can you get a backtrace[1] and file an issue[2] so we can take care of this? The information you've provided in this email would also be useful. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] Numbers hackers call
If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a premium service etc. (But someone jumped in with an explanation) I'm guessing you have nothing to add to the discussion? From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Steven Howes steve-li...@geekinter.net Sent: Wednesday, March 26, 2014 12:13 PM To: Asterisk Users List Subject: Re: [asterisk-users] Numbers hackers call On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... It's an international call to +972595XX, tried with the 00, 001 and no prefix What is confusing? Steve -- _ -- 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] Numbers hackers call
On 26 Mar 2014, at 16:20, Michelle Dupuis mdup...@ocg.ca wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a premium service etc. (But someone jumped in with an explanation) I never mentioned the 972 area code. It’s a country code - and as others have said it’s been mapped to a Palestinian mobile network. I’ve added this to my bar list - I’ve seen quite a lot of toll fraud to Palestine (and the middle east in general in recent months). If you’re referring to country code, then the 0 of the local number is dropped when dialled internationally, see: https://en.wikipedia.org/wiki/Telephone_numbers_in_Israel I'm guessing you have nothing to add to the discussion? Think what you will. Steve -- _ -- 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] Numbers hackers call
On 3/26/2014 12:20 PM, Michelle Dupuis wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a premium service etc. (But someone jumped in with an explanation) 0X or 0XX is only if you're in country and need to dial with the 0 national trunk code (much like dialing 1+ in the US for an in country but long distance call). Someone dialing from outside the country doesn't need to add the zero, so they just use the 972 country code + 59 prefix. -- _ -- 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] Strange dropped calls
Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless set. 5. The ISP reports a very clean circuit. (Ethernet from the CLEC.) 6. We don't see their phones become unavailable very often. 7. They are the only site that seems to be having trouble. So, where else can/should I look? Mike. -- _ -- 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] Strange dropped calls
I would suggest starting with a packet capture of the SIP messages that will include both call legs (i.e. capture at the Asterisk box). This should tell you who initiated the hangup - the carrier side, the phone side, or Asterisk. On Wed, Mar 26, 2014 at 11:46 AM, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless set. 5. The ISP reports a very clean circuit. (Ethernet from the CLEC.) 6. We don't see their phones become unavailable very often. 7. They are the only site that seems to be having trouble. So, where else can/should I look? Mike. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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
[asterisk-users] Default extension
Hello, When I get a SIP INVITE as follows: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18 To: sip:02XX@IP:5060 Contact: sip:1053212@IP:5060 Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Asterisk considers that the extension is 's'. (The Register) How to make the extension number that is shown in the 'To' ?? Thank you, Mickael -- _ -- 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] DAHDI-Linux and DAHDI-Tools v2.9.1-rc2 Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1-rc2 DAHDI-Tools-v2.9.1-rc2 dahdi-linux-complete-2.9.1-rc2+2.9.1-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release includes a firmware update for the wcaxx, wcte13xp, and wcte43x series of cards. This includes the A4B, A8B, TE133, TE131, WCTE235 and WCTE435. This firmware update resolves a rare case with certain chipsets that would cause transmitted audio to be muted for a short time on analog cards or spans to go down for a short time on digital cards. Additions from -rc1 xpp: fix PANIC for old dahdi_registration hotplug: call handle_device.d/ actions for remove registration-order: Added dahdi_auto_assign_compat Shortlog of dahdi-linux changes since v2.9.0: Oron Peled (1): xpp: fix PANIC for old dahdi_registration xpp: PRI stability fixes Shaun Ruffell (9): firmware: Refactor by using build_tools/install_firmware. build_tools/install_firwmare: Try to extract the .bin file from .tar.gz wcxb: Print running version when recommending power cycle. wcxb: Reset TDM engine on IO errors. wcxb: Add diagnostic message if DMA retries are increasing when DEBUG is defined. wcte13xp: Update firmware for TE133/TE131 to 780019 wcaxx: Update firmware for A8B/A4B to 1d0019/b0019. wcte43x: Update firmware for TE435 / TE235 to e0019. wcxb: Disable presence detect reporting on upstream port during PCIe hard reset. Tzafrir Cohen (1): dahdi_get_auto_assigned_spans Shortlog of dahdi-tools changes since v2.9.0.1: Aslan Laoz (1): waitfor_xpds: handle the case of a failing AB Shaun Ruffell (2): hotplug: Check for auto_assign_spans only when ACTION is add. dahdi_cfg: error()-perror() when sem_open fails. Tzafrir Cohen (1): auto_assign_spans may be true even if not '1' Oron Peled (2): hotplug: call handle_device.d/ actions for remove registration-order: Added dahdi_auto_assign_compat The diffstat from the dahdi-linux v2.9.0 release: build_tools/install_firmware| 23 + drivers/dahdi/dahdi-base.c | 12 ++- drivers/dahdi/firmware/Makefile | 191 +--- drivers/dahdi/wcaxx-base.c | 4 +- drivers/dahdi/wcte13xp-base.c | 13 ++- drivers/dahdi/wcte43x-base.c| 2 +- drivers/dahdi/wcxb.c| 97 drivers/dahdi/wcxb.h| 9 ++ drivers/dahdi/xpp/card_pri.c| 14 ++- drivers/dahdi/xpp/xbus-core.c | 2 +- drivers/dahdi/xpp/xpp_dahdi.c | 53 +++ drivers/dahdi/xpp/xpp_dahdi.h | 4 +- include/dahdi/kernel.h | 3 +- 13 files changed, 209 insertions(+), 218 deletions(-) The diffstat from the dahdi-tools v2.9.0.1 release: dahdi_cfg.c | 6 +++--- hotplug/dahdi_handle_device | 21 +++-- hotplug/dahdi_span_config | 25 ++--- xpp/dahdi_registration | 2 +- xpp/waitfor_xpds| 4 5 files changed, 33 insertions(+), 25 deletions(-) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.9.1-rc2 http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.9.1-rc2 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default extension
On Wed, Mar 26, 2014 at 1:14 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hello, When I get a SIP INVITE as follows: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18 To: sip:02XX@IP:5060 Contact: sip:1053212@IP:5060 Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Asterisk considers that the extension is 's'. (The Register) How to make the extension number that is shown in the 'To' ?? What version of Asterisk are you using? It would help to show how you are performing the dial in dialplan or otherwise. If you are dialing a user/peer present in sip.conf or a database then show that configuration as well. Based on that someone could make a suggestion. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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] Numbers hackers call
On Wed, 26 Mar 2014 16:20:58 + Michelle Dupuis mdup...@ocg.ca wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. You never dial the local trunk prefix when you're calling internationally. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Numbers hackers call
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace Sent: Wednesday, March 26, 2014 6:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Numbers hackers call If this is to 972 area code then the next digits should be 0X or 0XX but they are not. On Wed, 26 Mar 2014 16:20:58 + Michelle Dupuis mdup...@ocg.ca wrote: You never dial the local trunk prefix when you're calling internationally. Italy is the only exception where dialing from outside the country requires the leading 0, but that is because the leading 0 isn't used as a trunk prefix The ITU provides copies of each country's dialing plan for free at http://www.itu.int/oth/T0202.aspx?parent=T0202#A -- _ -- 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] (OT) Phones with STP, DHCP, and/or (T)FTP Issues
I remember having to turn off STP or set portfast on some switch ports to some phones due to the boot sequence and timeouts of some phones a long time ago. Does anyone know which phones, if any still suffer from these problems? I am setting up a lab and want to introduce this problem for the class. Thanks, Steve T -- _ -- 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] Security log format / content
I've noticed that the Asterisk (v11) security log captures attempts do dial without first authenticating, and places the number dialed into the accountid field. I'm trying to distinguish between failed attempts to register and attempts to dial without registering, but the security log treats them identically (using the accountid field for either the username or number dialed). I have noticed that the eventversion field is set to 2 for failed dial attempts, and 1 otherwise. Is this coincidence? Or can I rely on the eventversion=2 in the future to distinguish these two event types? (I've looked here: https://wiki.asterisk.org/wiki/display/AST/Security+Log+File+Format? but it doesn't really help) -- _ -- 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
