Re: [asterisk-users] Asterisk / PRI and Outbound Overlap Dialing (Mtt Cannon)
According to Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching#PatternMatching-SpecialCharactersUsedinPatternMatching The exclamation mark (!) character is similar to the period and matches zero or more remaining characters. It is used in overlap dialing to dial through Asterisk. For example, _9876! would match any number that began with 9876 including 9876, and would respond that the number was complete as soon as there was an unambiguous match. Asterisk treats a period or exclamation mark as the end of a pattern. If you want a period or exclamation mark in your pattern as a plain character you should put it into a character set: [.] or [!]. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1. SIP trunks going to the wrong context (Ade Vickers)
How about if you set; exten => _se,1,Dial(IAX2/cloud/1000,30,r) Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ooh323 - cisco call manager express (Dmitry Melekhov)
Is very hard to give a suggestion without more information, when call from CME to Asterisk no voice is detected on both path? how about to collect traffic information between Cisco an Asterisk? Without a call trace and analyzing with Cisco partner or someone with H323 experiences it will be very difficult to let you the causes of no voices. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1. asterisk 13.13.1 Everyone is busy-congested at this time
Hi Motty Cruz Probably it could become from missed configuration, check contexts issue. Check SIP context=sip-phone and extension dialplan context, probably you forget to set include or mistyping or other related to context issue. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI error: link goes down when make calls
Hi. These becomes from layer 1 issue and could becomes from a wide of possibilities (telco or Asterisk PRI card). At first check cables an connectors, on Asterisk could make a loopback and check if problem persist. On telco should check cable of DSL Modem or whatever provide PRI service. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic PBX connect to Asterisk (Ikka Tirtawidjaja)
Dear ikka Thanks, well the best way is by SIP to E1 ISDN gateway (could use Digium, Audiocodes, Sangoma, etc.). But should contemplated on KX-TDA600 ISDN PRI card KX-TDA0290 and configure as QSIG signalling. I don't recommend to use ATA FXS converter, first problem you couldn't dial to any extension to KX-TDA600, PBX answer incoming call from analogue line through Voice mail configured as Automated attendant, or DISA card. The other point is line attenuation. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic PBX connect to Asterisk (Ikka Tirtawidjaja)
Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 11.21.0 : echo woes : can't installcanceller (sean darcy)
Hi Sean In case of electrical I think DAHDI card if could have the capability by a combinations of jumpers it will be too hard to set best level, think about it for one or a wide device, could take long time for start-up, and will require to adjust or controller periodically, therefore many terminal devices could have it inside, by changing jumpers or some components to adapt according to electrical line condition By the other way market analogue terminals mainly follows, to FCC or European standards, I remember these kind of symptom happens with Panasonic KX-TS500, whatever PBX or line it feels echo. In case of acoustic analogue device it used analogue circuit, SIP used DSP (Digital signal processing circuit) these last one is better than analogue system. Mc GRATH Ricardo -- _ -- 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] 11.21.0 : echo woes : can't install canceller (sean)
Hi sean Sorry echo canceller it's only works for FXO ports (PSTN Line). Mc GRATH Ricardo -- _ -- 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] 11.21.0 : echo woes : can't install canceller (sean darcy)
Hi Sean Darcy Question about "the remote party always hears an echo on it's side", strange because eco suppression circuit is for local side. Mc GRATH Ricardo -- _ -- 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 handle multiple lines call
Hi Thyda
When you set "exten => _.,1,Dial(SIP/${EXTEN})" Asterisk assume "_.", an match
everything on your dialplan including special extensions as "i", "h", etc.,
these will cause problems onto your system.
If you need to match one or more digits you can use _x and _x.
_x it mean only one pattern digit form 0 to 9
_x. any pattern digit from 0 to 9 and dot it mean remnant digits could be 2 or
3, 4 ... etc., so what ever you dial on sip it will be valid.
Mc GRATH Ricardo
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Re: [asterisk-users] Help With Physical Layer (Tony Kasule)
Tim At first should take a look to cable pinout (RAD documents) as pin 1,2, Transmit (output) and 4, 5 Receive (input) for Digium card you should use a straight cable (try to test with new cable one too). Second check Dahdi configuration parameters, use dahdi commands as; dahdi show status, service dahdi restart and check result, (could a mistake on parameter value on system.conf). Mc GRATH Ricardo -- _ -- 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] Connecting Asterisk and BT Versatility PBX via NT BRI
Roberto Thanks. Panasonic KX-TD1232 system ISDN interface use DDK plug, not same cord you are using for openvox B400P card. Each port could be configured as CO or extension port as Point-Point or PTMP in both mode. Regarding to dial timeout these it becomes from ISDN dial mode overlap or En-bloc, by the way you can close dialling number by using # key. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] Connecting Asterisk and BT Versatility PBX via NT BRI
Roberto Could you provide more details about Panasonic PBX test? model unit and configuration details? Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] Connecting Asterisk and BT Versatility PBX via NT BRI port
Hi, it seems have problem with physical issue layer 1, first check wiring connection, by the way you can check with card led (If ISDN plugs into the port, the LED will not blink, but in red color.) after dahdi alarm status, and dahdi restart command. At last should be signalling mode to signalling = bri_cpe_ptmp Good luck Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] T1 Card RED ALARM
Why you configure zaptel.conf? should configure on dahdi files Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP
It seems a layer 2 problem, should check about references point (network or terminal equipment), it assume your Asterisk box is connected to ISDN PSTN provided, just in case check at you side if all related configuration files are configured as signalling=pri_cpe (Card config, wan_cfg, chan_dahdi.conf), PSTN side if network configuration or in service mode, should both side work and debug in the same time. Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] Use Allworx Phones With Vanila Asterisk PBX? (Jr Richardson)
Hi It seems these device it works similar way as a Cisco phone, so no way to look how to configure the phone because is based on template files downloaded from main system, files are downloading by TFTP option #66 (new software configurations file etc.) when device boot up. If user need to configures directory and others "My Allworx manager " it should made through system ip address or domain name. I think better idea to choose another brand, save your time! good luck. Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] 3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera)
Hi Leonardo At first should be useful to post your message at asterisk-r2-requ...@lists.digium.com group. By the other way let me advice, to make an explained detail of your problem as; Asterisk version Openr2 version Configurations files Dialplan dahdi pattern detail Detail of the call process (inbound or outbound call failed?) In case of outbound call failed extension dial wait time of the call process (waiting for ringback tone, busy, reorder, silence etc) call trace log (for better collect information, you could move or delete old data log on /var/log asterisk/.) In case of inbound call failed, it could be possible with call log and monitoring bit CAS signalling Based on your explanation and information could let you know the possible reason of the weird issue. By the other way, let me explain when dahdi restart, it make hardware reset, therefore will stop card control software and restart again, these will cause lost frame connection between you box and PSTN, and call process, etc. When dahdi restart again it recover frame between both side and set CAS control channel according to dahdi parameters setting on system.conf These is for preventing to let to PSTN start call handler process during asterisk starting process or stop. Another point it seems on your log information a call have been answered, so it seems is working. Good luck PD: 0x00 it doesn't mean idle state (it mean force release) Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] trunking trixbox - panasonic
Hi could let more details of your test scenery? Just in case of Panasonic, if you use SIP trunk resource, it need to configured on CO Lines settings (DIL Tables) distribution method and DDI tables, for incoming calls. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] How to configure NT/ptmp with Dahdi and BRI ? (Olivier)
Hi Olivier It seems wrong configuration, because according to your mail Asterisk it will be acting as terminal mode (ie Patton gateway acting as network and asterisk as terminal). But Asterisk message is indicated Asterisk a s a NT mode [ 212.226050] wctdm24xxp :0a:01.0: xhfc: Configuring port 0 span 1 in NT mode with termination resistance ENABLED. It could help by checking parameters on dahdi-channels if signalling = bri_cpe_ptmp # The signalling for TE mode Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] Recommendations for SIP to ISDN PRI E1 gateway to use with
Hi John How about Digium G100 G200 voip gateways? You can try with a test drive at Digium web site http://www1.digium.com/en/products/voip-gateways Best regards. Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] How to remove the call waiting tone without disabling
Niccol How about to change tone list on indications.conf file? Please comment call waiting line ";" according to country zone or default settings. Good luck Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] Caller ID : FSK ETSI or FSK US
Dear In essence Caller ID ETSI and FSK US (Bellcore) is based on the same pattern as; _ ___ _ _ |First Ring burst |_500ms_|Channel seizure 300 bits|__|Mark Signal|__| Caller ID Message|_200 ms_|Second ring burst | So basically any kind of device should be work without any problem, unfortunately during these process if some noises (as miss ground connection or others) happens during the process can make failed to process caller-id information, by the modem. Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] Asterisk - Nortel transfer problem
Hi Carlos It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk. I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchange DR2 signalling between Nortel and Asterisk is about 5 sec. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] Asterisk - Nortel transfer problem
Hi Carlos How about to perform the call transfer operation (screened and unscreened) and trace the communication it will better on Nortel PBX side, in order to know which point is causing call transfer failure. By the way these kind of features is complicated to handler on interconnected system, I just mean is no easy way for check intersystem resource state, as other than inband tones, mainly PBX system interconnect to others system by trunk resource and communication between system are handler by Trunk to Trunk PBX system control operation, PBX only use remote state (Busy, Congestion, etc.) for management own Trunk to Trunk resource operation. Just in case of these kind of scenario it can be done through E1 (DR2 MFCR2 or DTMF) or ISDN PRI QSIG, lastone is more convenience because it cover more supplementary services than DR2 as; Completion of Calls to BusySubscriber (CCBS), Call Hold (HOLD)—by ISDN etc. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)
Hi I'm not agree problem could be cause from IRQ setting, I think in that way problem should be more unstable, moreover no voice communication problem with DAHDI service start up. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)
Hi Anam It seem should be work, but I just have a question about chan_dahdi.conf regardless to parameter rxwink=300 ; Atlas seems to use long (250ms) winks By the way gain parameters shouldn´t have any effect to CID signal processing, how about to comment, and test again, if still without working try to connect parallel phone with Asterisk it will check if could be a hardware problem or configuration parameter setting Good luck Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] dahdi cannot make simaltaneous calls
Hi rosli sukri Well what I see release it becomes from outbound PRI server (PSTN or whatever). By the way it seems according to the PRI trace first call label 32771 have an uncompleted process, voice communication isn´t established (Answer connect acknowledge etc message), called party ringer, and answer? In case of 32772 outbound PRI server side it no response to Setup request, moreover after send Setup message it response with a release message cause Network Congestion (resource unavailable). So it can conclude outbound system ???-->dahdi -->asterisk,-->Sip can´t process simultaneous call. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.com From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com [asterisk-users-requ...@lists.digium.com] Sent: 20 April 2012 10:22 To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 93, Issue 28 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: g729 freezes 1.8 (A J Stiles) 2. Re: g729 freezes 1.8 (Kevin P. Fleming) 3. dahdi cannot make simaltaneous calls (rosli sukri) 4. Company info (Josu? Conti) 5. Re: Company info (Josu? Conti) 6. Auto answer Asterisk ;Unable to create channel of type (motty.cruz) 7. Re: g729 freezes 1.8 (Jeff Brower) 8. Re: asterisk 1.4.39 and dahdi 2.6 on Ubuntu (bilal ghayyad) 9. Re: asterisk 1.4.39 and dahdi 2.6 (Chad Wallace) 10. Re: Company info (Steven Howes) 11. Re: Company info (Doug Lytle) 12. Re: Company info (Josu? Conti) 13. Experience with virtual servers? (Brynjolfur Thorvardsson) 14. Re: Experience with virtual servers? (Arthur Stanfield) 15. Re: Experience with virtual servers? (Danny Nicholas) 16. Re: Experience with virtual servers? (Stuart Elvish - IP Exchange Systems) -- Message: 1 Date: Thu, 19 Apr 2012 17:52:17 +0100 From: A J Stiles Subject: Re: [asterisk-users] g729 freezes 1.8 To: "Asterisk Users Mailing List - Non-Commercial Discussion" Message-ID: <201204191752.17519.asterisk_l...@earthshod.co.uk> Content-Type: Text/Plain; charset="iso-8859-6" On Thursday 19 April 2012, samuel wrote: > Just in case it helps: > > It turned out that from asterisk version 1.8.4 on, the g729 binaries are > different from the previous versions so it was a version mismatch between > the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher). > > Thanks to the Digium support department that found out the issue. Someone really needs to get the mPlayer folks (based on the Continent, where mathematics is not patentable) to create an Open Source g729 codec implementation . -- AJS Answers come *after* questions. -- Message: 2 Date: Thu, 19 Apr 2012 12:59:56 -0500 From: "Kevin P. Fleming" Subject: Re: [asterisk-users] g729 freezes 1.8 To: asterisk-users@lists.digium.com Message-ID: <4f90529c.8070...@digium.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 04/19/2012 11:52 AM, A J Stiles wrote: > On Thursday 19 April 2012, samuel wrote: >> Just in case it helps: >> >> It turned out that from asterisk version 1.8.4 on, the g729 binaries are >> different from the previous versions so it was a version mismatch between >> the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher). >> >> Thanks to the Digium support department that found out the issue. > > Someone really needs to get the mPlayer folks (based on the Continent, where > mathematics is not patentable) to create an Open Source g729 codec > implementation . Source code availability is not the issue; the reference source code is easily obtained from the ITU-T. Many of the G.729 patent holders are companies based in Europe, so I suspect they would have a different opinion than you do about the legitimacy of their patent claims on G.729 :-) In any case (and of course IANAL), it is my understanding that the patents that cover the base G.729 recommendation, along with Appendices A and B, will all expire in the next year or so. We'll have to see what that means for the market, especially with new, more freely licensed, codecs coming out that provide substantially better performance. -- Kevi
[asterisk-users] Low cost BRI gateway
Dear Chris How about to use 2 Asterisk system interconnected through Wireless solution point to point one system should be for ISDN BRI gateway with Digium PCI card and the other server for extension voice mal and so on. for covering distances it will better to use Motorola Canopy web site or Airmax from Ubiquiti (web site http://www.ubnt.com/airmax). It should be as the following (-<><><><><><><><><><><> -) PSTN/ISDN BRI Lines <-BRI-> Asterisk<-LAN-> /\ Wireless point to point /\ <->LAN Asterisk customer side registering handsets, IVRs, voicemail, etc In general these system operates on 900 MHz 2.4 GHz, 5.2 GHz, 5.4 GHz, and 5.7 GHz and can cover a long distances depending to environment. Please be advise to check local regulation about using wireless system (frequency operation range, authorization and others. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] ISDN, overlap and open dialing plans (Olivier)
Dear Oliver Well I never knew PBX could according to numbers pattern length use enbloc and overlap, from my experiences dialling mode setting should be one or other, but it should be set on whole system one mode, by the way if number length pattern is variable component to use overlap mode. Just in case when it use enbloc PBX send whole number, TDM phones it press # or other key setting as send digits. By the other way check dialplan rules to resolve receiving number length, a good practice is use and Asterisk extension to simulated call from PBX system. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.com From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com [asterisk-users-requ...@lists.digium.com] Sent: 13 March 2012 14:00 To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 92, Issue 21 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. ISDN, overlap and open dialing plans (Olivier) 2. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) (rama...@gmx.de) 3. Re: Capacity of single instance of Asterisk (Amit Patkar | Avhan Technologies Pvt Ltd) 4. Re: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) (Eric Wieling) 5. Re: Capacity of single instance of Asterisk (Amit Patkar | Avhan Technologies Pvt Ltd) 6. Re: how to show used "wrong password" (Kevin P. Fleming) 7. Re: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) (Shaun Ruffell) 8. Re: Capacity of single instance of Asterisk (Kevin P. Fleming) 9. Re: Capacity of single instance of Asterisk (Bryant Zimmerman) 10. Re: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) (Tzafrir Cohen) 11. Re: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) (rama...@gmx.de) 12. Re: how to show used "wrong password" (Randall) 13. Re: how to show used "wrong password" (Randall) -- Message: 1 Date: Tue, 13 Mar 2012 14:37:06 +0100 From: Olivier Subject: [asterisk-users] ISDN, overlap and open dialing plans To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: Content-Type: text/plain; charset=ISO-8859-1 Hi, I've got the following setup: PSTN/ISDN < E1-> Asterisk < E1-> Alcatel 4400 PBX <> TDM phones When a TDM phone is dialing out to a national number, it seems that the PBX is using enbloc dialing. When a TDM phone is dialing out to an international number (variable length numbers), it seems that the PBX is using overlap dialing as Asterisk is currently receiving truncated numbers. What is the best way to deal with such situations ? 1. configure PSTN in enbloc dialing and tweak dialplan to mimic overlap dialing ? 2. or configure both PTSN and PBX spans in overlap mode ? Suggestions ? Regards -- Message: 2 Date: Tue, 13 Mar 2012 15:30:45 +0100 From: rama...@gmx.de Subject: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) To: asterisk-users@lists.digium.com Message-ID: <20120313143045.18...@gmx.net> Content-Type: text/plain; charset="utf-8" Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Thanks Rainer Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, m
[asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link (Dustin fails)
Can be done calls from each system? How about to capture data with Wireshark? I have experiences Asterisk with Panasonic with H323 without any problem. Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
