[asterisk-users] NIN Ghosts music (free download) safe for MOH?
Is the new NIN Ghosts music (free download) safe for MOH? Justin Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Pickup
>We have a customer service queue which works great. The members are hard >coded (member => SIP/1000), etc. However, we have a special need. If the >queue becomes busy, we would like to be able to dial an extension and >grab only the next caller in the queue. We don't want to log in as an >agent, since that would add another step (logging in/logging out). I saw >there was a Pickup() command, but I'm not sure if this will work with >queues. I have a "reverse transfer" module I wrote. I could probably adapt this for queues without too much work. Thoughts? Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broadcast/Announce app
Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there is any interest and I'll post the code. Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk trunk/1.6 and nvfaxdetect
I'll begin working on full cross-version support (Asterisk 1.2, 1.4, and 1.6) in early May for nvfaxdetect and a handful of other modules. Justin Newman > >Hi, > >we are using the app_nvfaxdetect from Newman Telecom with Asterisk 1.4 >and tried to build the trunk/next release 1.6 with this application, but >it failed (We are using fax stuff with iaxmodem/Hylafax). > >I remember that we had the same issue switching from 1.2 to 1.4 and >someone made the port (We don't have the necessary knowledge to do it). > >Has anyone port this application to last trunk and would share the port? > Or is the native Asterisk fax feature (spandsp) stable enough to >replace faxdetect? > >Regards > >-- >Daniel >TOOTAi Networks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP KEEPALIVES, with QUALIFY and without making UNREACHABLE
Need SIP KEEPALIVES in Asterisk, but QUALIFY won't presently work for you (due to it's channel disabling behavior)? Someone posted on the list that they would like to split "keepalives" and "qualify" into different features. Sounds like a good plan, but until that is done you can turn "qualify=" into a keepalive mechanism, without disabling your channels. Here's a quick fix: 1) Open "chan_sip.c". 2) Replace "lastms = -1" with "lastms = 0". 3) Save. 4) #make 5) #make install I've used it in the past without problems. Not perfect (or even close), but it works. Justin __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
Did this just start happening with the 1.4 tree? Have you made any progress on getting it resolved? Justin Newman >Matt Riddell wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Tzafrir Cohen wrote: >>> Let's be more specific here, folks: >>> >>> What version numbers? >>> >>> Asterisk, spandsp, agx-addons / rx-tx-fax? >> >> Asterisk: yesterday's 1.4 SVN >> SpanDSP: tried with pre 15, 16 and 18 >> AGX-Addons: tried with 1.4.5 and svn trunk >> rx/txfax: supplied by AGX Addons - although they seem to build the files >> and stick them into the modules directory, rather than adding to the >> apps directory and modifying the Makefile. > >i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 >linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax >when i enable faxdetect in zapata.conf. since then it disabled >faxdetect and use nvfaxdetect function in dialplan, it works >fine afterward. > >also it seems to works fine using regular 32bit kernel. > >-- >Edwin Lam <[EMAIL PROTECTED]> >Systems Engineer, Office General, Inc. >Ph: +1 415 439 4988 Fax: +1 415 283 3370 >http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need good voicemail documentation
Dave, Docos for Comedian Mail? Justin >From: dave cantera <[EMAIL PROTECTED]> >Subject: Re: [asterisk-users] Need good voicemail documentation > >An HTML attachment was scrubbed... >URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20080208/501668f8/attachment.htm __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nvfaxdetect, nvvoicemail, and others
I'll open the source repository soon for envy and nv suite of tools, including nvfaxdetect. I have a few handfuls of useful Asterisk add-ons. Starting on module updates to fully support Asterisk 1.2, 1.4, and 1.6 in May. Maybe we can get some of these in agx-ast-addons. Also, I am interested to see how the 3rd party tools community develops. Justin Newman >Hi Justin, > >On Thu, 2007-12-27 at 15:38 -0800, Justin Newman wrote: >> Yes, I wrote nvfaxdetect and a number of other modules. I don't have >> any nvfaxdetect updates planned for public release unless someone >> would like to integrate some of my changes in the GPL version...we >> could do this though. > >Perhaps you could send the diff to Antonio Gallo who started the >agx-ast-addons project which includes faxdetect and backgrounddetect >ported to 1.4. He seems open to enhancements/additions. His email is >agx at users.sourceforge.net The project can be found at: >http://sourceforge.net/projects/agx-ast-addons >http://agx-ast-addons.svn.sourceforge.net/viewvc/agx-ast-addons/trunk/ > >Regards, >Patrick > >> - Original Message >> From: Matt Riddell <[EMAIL PROTECTED]> >> >> Justin Newman wrote: >> > We just completed a new implementation of voicemail for Asterisk. >> It's much cleaner than Comedian mail and can emulate several voicemail >> user interfaces, including Audix. It's a great replacement for Audix. >> All of the sounds/prompts are presently being re-recorded by a >> professional female voice. >> >> Also, are you the guy who wrote nvfaxdetect et al? >> >> Any chance of an update for 1.4 etc? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
Me too. I have a Java client which works with Asterisk and Salesforce... Justin -- From: Olivier <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Asterisk - CRM Integration To me, "CRM-Asterisk integration" has several meanings. It could refer to : - basic click2call feature from CRM contact or project panel, - journaling Asterisk incoming and outgoing calls inside CRM projects data, - programming and executing Conference calls defined inside CRM projects data - screen popup - etc... Which feature are you specifically looking for ? Do you plan to use it in a call center or casual business office ? Regards Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New voicemail app (supports many interfaces, including Audix)
We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
Yes, I wrote nvfaxdetect and a number of other modules. I don't have any nvfaxdetect updates planned for public release unless someone would like to integrate some of my changes in the GPL version...we could do this though. - Original Message From: Matt Riddell <[EMAIL PROTECTED]> Justin Newman wrote: > We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. Also, are you the guy who wrote nvfaxdetect et al? Any chance of an update for 1.4 etc? Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
It's licensed GPL. I'm working on getting the web-site, documentation, and packaging up to par... if you're interested in helping, let me know. Here are some details on it: * Written for Asterisk 1.4.x; not tested with prior versions * Supports both voice and fax mail (including fax detection) * Database support build-in; can use real time as well * Web-based GUI for basic management * Professional non-Allison female prompts (English due mid-Jan 2008) * Consolidated MWI server/client comes with it (for consolidated or distributed voicemail servers) Justin - Original Message From: Matt Riddell <[EMAIL PROTECTED]> To: Justin Newman <[EMAIL PROTECTED]>; Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Thursday, December 27, 2007 3:08:31 PM Subject: Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix) -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Newman wrote: > We just completed a new implementation of voicemail for Asterisk. It's much > cleaner than Comedian mail and can emulate several voicemail user interfaces, > including Audix. It's a great replacement for Audix. All of the > sounds/prompts are presently being re-recorded by a professional female voice. > > If you are interest in the app, let us know at [EMAIL PROTECTED] I'm assuming that since you sent it to Asterisk Users (Non-Commercial Discussion) it is free. Is it also Open Source? What licence? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHdDBvDQNt8rg0Kp4RArv+AJ43NV5Rtxtx5+nuLf9kOclIOBRuwwCgnuM0 VK4Mg+svmfczGsffotPe24w= =CcGs -END PGP SIGNATURE- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail vs. minivm
This system targets a different market... I like Olle's system. He did a good job. Olle's minivm is a great choice for those wishing to build customized voicemail systems, but as the name suggests, the systems are very basic. Large systems are difficult to maintain in the dial plan and some of the functionality we need would be difficult to implement with that approach. Justin - Original Message From: Tzafrir Cohen <[EMAIL PROTECTED]> To: Justin Newman <[EMAIL PROTECTED]> Sent: Friday, December 28, 2007 2:29:28 PM Subject: Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix) Hi On Fri, Dec 28, 2007 at 02:19:35PM -0800, Justin Newman wrote: > It's licensed GPL. I'm working on getting the web-site, documentation, and > packaging up to par... if you're interested in helping, let me know. > > Here are some details on it: > > * Written for Asterisk 1.4.x; not tested with prior versions > * Supports both voice and fax mail (including fax detection) > * Database support build-in; can use real time as well > * Web-based GUI for basic management > * Professional non-Allison female prompts (English due mid-Jan 2008) > * Consolidated MWI server/client comes with it (for consolidated or > distributed voicemail servers) Again: did you get a chance to look at Olle's minivm? He generally broke down the voicemail functionality to separate apps that could be included in the dialplan to make a voicemail menu. IIRC it lacks support for other backends that app_voicemail currently has. But it is probably much less as messy. I wonder if it won't be a better base for extending. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Building prototype devices?
I know a lot of people on this list are building devices and equipment for Asterisk and communications in general... For those of you building prototype devices, you may want to check out TechShop in the bay area. They are expanding all over the place. http://www.techshop.ws They have lasers, etchers, welders, 3d shaping machines, lathes, and a bunch of other fancy equipment for making equipment prototypes... Justin Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail App
Red Tiger is Java based, so it will run on any Java VM (i.e., Windows, MacOS, Linux, Unix, etc.) There are some JNI-based additions for Linux which give it more capabilities, but Red Tiger itself runs cross platform. You could run Asterisk on Linux, but have Red Tiger and all of your applications running on Windows, MacOS, Linux, etc. Only the base libraries must be on the Asterisk machine... the rest can run anywhere. - Original Message From: Tammy A. Wisdom <[EMAIL PROTECTED]> To: Justin Newman <[EMAIL PROTECTED]> Sent: Friday, December 28, 2007 3:51:34 PM Subject: Re: Voicemail App What platform does red tiger run on? Thanks --Tammy Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail systems- flow charts, digit/key cards, etc
Does anyone have flow charts or digit/key cards for some of the more popular voicemail systems out there? (shows which digits/keys to press, where it takes you, etc.) I need to create some of the new voicemail system. Send 'em my way if you have them. nt_jnewman at yahoo.com Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc
We've implemented the Audix-similar system. I'd welcome a bounty! ;) Please send over cards for the other two systems you mentioned... - Original Message From: Andrew Joakimsen <[EMAIL PROTECTED]> To: Justin Newman <[EMAIL PROTECTED]>; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 16, 2008 9:05:40 PM Subject: Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc Oh and here's Audix: http://mx.netjdn.com/manuals/Legacy/quickref.pdf Heck I'd think I'd put out a bounty on making Asterisk voicemail like Audix if there were some people interested. On Jan 16, 2008 11:27 PM, Justin Newman <[EMAIL PROTECTED]> wrote: > Does anyone have flow charts or digit/key cards for some of the more popular > voicemail systems out there? > (shows which digits/keys to press, where it takes you, etc.) > > I need to create some of the new voicemail system. > > Send 'em my way if you have them. > > nt_jnewman at yahoo.com > > Justin > > > > > Looking for last minute shopping deals? > Find them fast with Yahoo! Search. > http://tools.search.yahoo.com/newsearch/category.php?category=shopping > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc
Let me see how much work it would be...might be able to include those as well. - Original Message From: Steve Totaro <[EMAIL PROTECTED]> To: Justin Newman <[EMAIL PROTECTED]>; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 16, 2008 8:43:28 PM Subject: Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc 3Com http://www.sjc.cc.nm.us/documents/ots/docs/VoiceMailGuide.pdf NEC Elitemail http://gigshowcase.com/EndUserFiles/2912.pdf A system similar to Elitemail would rock! Thanks, Steve Totaro On Jan 16, 2008 11:27 PM, Justin Newman <[EMAIL PROTECTED] > wrote: Does anyone have flow charts or digit/key cards for some of the more popular voicemail systems out there? (shows which digits/keys to press, where it takes you, etc.) I need to create some of the new voicemail system. Send 'em my way if you have them. nt_jnewman at yahoo.com Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More voicemail cards needed...
Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * AT&T/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on getting these integrated with EVM. Users will be able to select via user prefs and admin on a per user setting of their preferred VM flow. Final prompts are coming this week; need the cards for any additions. I have the following: Audix, 3COM, IC IMail, NEC EliteMail LX, Nortel Call Pilot, Olle's, and a customized Octel. Feel free to send others that may be of interest. Send all cards to: nt_jnewman at yahoo.com. Justin Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for unified messaging expert
Anyone in the bay area with strong unified messaging experience? Respond off list at: [EMAIL PROTECTED] Justin Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling NVFaxDetect and other Newman apps on Asterisk 1.4
If you are having problems compiling NVFaxDetect (app_nv_faxdetect.c) or other Newman Telecom applications on Asterisk 1.4, please look at Steve's comments at: http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14 Several changes to Asterisk prevents NVFaxDetect and other apps from registering. Some changes needed. He also has copies of the code if you need it... Justin Newman Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Faxing Support
We have considered working on this. T38 is a short term solution, though. Justin Newman -- From: Tomislav Par?ina <[EMAIL PROTECTED]> Subject: [asterisk-users] Re: Asterisk Faxing Support In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Asterisk 1.2 has no support of t.38 whatsoever, the call will drop > before t.38 is ever utilised, not even pass-thru. > > 1.4 Adds support for T.38 pass through only and no other sort of > faxing, the endpoint must support T.38 and you must send your call to > a T.38 gateway and you must not use NAT anywhere in your network and > you must enable re-invites which could cause CDRs not to reflect the > true details of the call. > > Asterisk/Digium also has no interest in any further interest in > expanding T.38 or faxing support in Asterisk. > > Steve Underwood and the other fine persons that have helped to develop > the software DSPs and other stuff required for FoIP support also have > no interest in writing any further faxing support for Asterisk (RxFax, > TxFax + the newest span_dsp wont even compile, much less work under > Asterisk any more) probably because they know it will never be > included into the Asterisk code. Someone please tell me this isn't truth. Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Auto Answer (Paging)
What kind of phones? Polycom? You could also use one of the sip-based speakers or intercom units... -- From: Rob Schall <[EMAIL PROTECTED]> Subject: [asterisk-users] Auto Answer (Paging) I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to hear that meeting, but instead the people in the meeting could hear the "hellohello", and then that's it. Is it possible to have a auto-muted auto-pickup call? Rob Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Pickup application
Try NVPickup.
--
Date: Fri, 16 Feb 2007 09:24:08 +0100
From: nik600 <[EMAIL PROTECTED]>
Subject: [asterisk-users] Pickup application
I am trying to configure the pickup.
This is my dialplan:
exten => _57.,1,Pickup(${EXTEN:2})
So, when i call for example 57333 Asterisk tries to pick up the call
ringing on 333
The problem is that it works only with internal calls!
For example, if i call 333 from 334 and while 333 i ringing i try to
dial 57333 it works.
If i call an external number that via dialplan dials 333 dialing 57333 i got:
Executing [EMAIL PROTECTED]:1] Pickup("SIP/200-08432290", "333") in new stack
[Feb 16 09:20:46] NOTICE[7586]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 333.
Where am i wrong?
Thanks
Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
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[asterisk-users] Re: Passing a variable from one Asterisk box to another
Use SIP headers... you could also use the SIP headers to store a ID/key for lookups against a database (e.g., store your call associated data in the database, for use by multiple machines). Justin Newman Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/features_spam.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please help test the gender detection module at 575-613-4392
Tzafrir, What other information do you think we could collect during the test that would be useful sorting through the data? Justin --- Justin Newman Sr Software Engineer Envy Software LLC nt_asterisk at yahoo.com (general list) nt_jnewman at yahoo.com (personal) justin_newman (skype im) justin_newman (yahoo im) From: Tzafrir Cohen To: asterisk-users@lists.digium.com Sent: Saturday, February 21, 2009 11:42:41 AM Subject: Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 On Sat, Feb 21, 2009 at 09:35:46AM -0800, Asterisk Asterisk wrote: > I'd also like to collect more information, including age and zip. > Figuring out how to do these things without affecting the tests > seems to take more time than the code itself. ZIP would be meaningful to you if the caller is from the US. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Update: Looks like Nufone is finally dead and out of business
Another one bites the dust: Looks like Nufone is finally dead and out of business. I heard they may be selling some of their software, name, and other items to another company, but that it probably won't happen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 21
-- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson <[EMAIL PROTECTED]> Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Friends in the Asterisk community, One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I've constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I've been kind of avoiding during my trainings is voicemail. And there's where I've spent a lot of time recently... Life is strange. Instead of fixing the current voicemail, I decided to restart. Break up large apps into small building blocks, allowing Asterisk admins to use the rich dialplan script language or AEL to build a voicemail solution that fits the organization. I've named this minivoicemail, which for each addition becomes more of a bad choice of name for this project. Flexivoicemail could be better... :-) I've removed functionality like ODBC and IMAP support, something that can be reapplied later. I've also not replaced the hooks into other channels for voicemail notification, but that can be done too. I haven't started replacing voicemailmain(), since I've focused on the need of larger systems where one only supports e-mail notifications of voicemail with audio attached. What I currently have is: Applications - MinivmGreetPlay voicemail greetings (busy/unavailable/temporary) - MinivmRecordRecord voicemail message - MinivmNotifyNotify account owner of message (email, pager) - MinivmDeleteDelete message Functions - MINIVMACCOUNT() - Get properties of voicemail account CLI commands - minivm show settings - minivm reload - minivm show stats - minivm list accounts - minivm list templates New features: - I've added support for e-mail and pager templates in various languages. - All apps are usable without setting up a voicemail "account" for a user. Just run the app with an e-mail address as an argument. The branch is based on Asterisk 1.2 and can easily be downloaded from http://svn.digium.com/svn/asterisk/team/oej/minivoicemail I need testers, ideas for new applications and possibly coders that can help to complete this. To start - Checkout this branch, compile and install - Check the minivm.conf.sample for instructions - Read the top of the source code file for ideas, todo's and changes http://svn.digium.com/view/asterisk/team/oej/minivoicemail/apps/ app_minivm.c?view=markup (And if you want to encourage me further, paypal to [EMAIL PROTECTED], thanks) Thanks for your help building a more flexible voicemail system for Asterisk! Send bug reports, comments and ideas directly to me and I'll try to summarize. /Olle Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Coaching in asterisk
NVWhisper. Justin -- Date: Thu, 08 Mar 2007 16:25:28 -0500 From: Wai Wu <[EMAIL PROTECTED]> Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx TV dinner still cooling? Check out "Tonight's Picks" on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Recorded file processing app wanted
There are plenty of apps. Audicity is one of them, which is multi-platform oss. You can script or batch process all your files... Let me know if you need help with this. I can show you how. Justin Newman -- Date: Fri, 9 Mar 2007 13:16:39 -0800 (PST) From: Steve Edwards <[EMAIL PROTECTED]> Subject: [asterisk-users] Recorded file processing app wanted Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NVFaxEmail
> Date: Fri, 08 Apr 2005 09:20:26 +0200 > From: Chris Blake <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Fax to email problem > To: Guy Decarpentrie <[EMAIL PROTECTED]> > > On Thu, 2005-04-07 at 17:50, Guy Decarpentrie wrote: > > Le jeudi 7 Avril 2005 16:43, Chris Blake a écrit : > > > Greetings *`s, > > > > > > I am trying to get faxes rec`d by * to be passed over to an email > > > address, and although the fax is being rec`d, it is not being > > > transmitted to the email address : > > > > > Apr 7 18:07:24 WARNING[2078]: Unable to execute 'mime-construct --to > > > [EMAIL PROTECTED] --subject "Fax from 0 " --attachment 0.pdf --type > > > application/pdf --file /var/spool/asterisk/fax/1112889947.49.tif.pdf' > > > --- > > > > Are you sure that you've installed the mime-construct package ? > > Use NVFaxEmail... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: external access to voicemail?
> Date: Fri, 8 Apr 2005 16:21:03 +0900 > From: "Mick Hastings" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] external access to voicemail? > > Hi all, > > I currently have a setup where my users dial in to a dedicated DID that > sends them to VoiceMailMain(). this works fine except for the fact that > nobody can remember the number! (they already have to remember the main > number, their personal number, fax number and mobile number) > > What I would like to setup is a way of people checking there own voicemail > by dialing there normal extension DID, waiting for it to go to VoiceMail() > and then keying in a secret code (or maybe just * as they are required to > enter a password later anyway) that switches them to VoiceMailMain() for > checking their messages. > > Has anyone already done this? I know it is quite common on home answering > machines. > > I guess its just a matter of checking for DTMF whilst playing back the > unavailable message or something? Can this be done without being integrated > into the VoiceMail() code? > > cheers for all the help, > Mick In our setup, we allow the user to press "#" to access their voicemail messages (voicemailmain)... if you need help, email [EMAIL PROTECTED] and we'll walk you through it. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: X100P doesn't check for dialtone
> Date: Fri, 8 Apr 2005 09:49:17 -0400 > From: "Malcolm Taylor" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] X100P doesn't check for dialtone > To: > > I have connected my home phone line into my asterisk box via an X100P, but > have noticed that asterisk doesn't check the line for dialtone before > dialing, barging in on any non-asterisk call which is taking place. > > > > I see from the voip-info.org wishlist that there is an outstanding item to > 'Listen for dial tone before dialing' and I also see someone suggesting a > solution for the problem by adding additional hardware into the home phone > circuit. > > > > I'm just wondering if anyone can recommend another (perhaps > zapata.conf-based) fix. Search for NVLineDetect on the wiki or on Google. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Fax to Email
> Date: Sun, 10 Apr 2005 11:06:59 -0500 > From: Bill Ford <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Fax to Email > > This has already been answered...but I can't find it... > > Has anyone set up multiple fax lines in asterisk... > > Fax Extension #1 goes to email1 > Fax Extension #2 goes to email2 > ETC... > > In other words, I want to be able to give numerous users each > a "virtual" fax machine.. > > Bill ; Assumes entry is DID # or extension number [context-incoming] exten => some_did,1,NVFaxEmail([EMAIL PROTECTED],Someone) exten => some_other_did,1,NVFaxEmail([EMAIL PROTECTED],Someone2) You could use NVFaxDetect first to check for the presence of the fax. This sample requires SpanDSP and NVFaxEmail. Alternatively, you could use SpanDSP, RxFax, and a different AGI script or app. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Email to Fax
> Message: 11 > Date: Thu, 21 Apr 2005 20:39:22 -0500 > From: "Anton Krall" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Email to Fax > > Anybody doing email to fax using spandsp? > Yep... Justin Newman Newman Telecom, Inc. [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: RJ45 to RJ11?
If you need to, just go to Radio Shack and buy the 2 RJ-11 to RJ-45 adapter. They're $5.99. Justin Newman Newman Telecom, Inc. --- Message: 29 Date: Wed, 27 Apr 2005 14:40:26 -0400 (EDT) From: "Jon Pounder" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] RJ45 to RJ11? To: "Asterisk Users Mailing List - Non-Commercial Discussion" Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain;charset=iso-8859-1 > I just received my TDM400 card from digium with 2 fxo and 2 fxs > interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS > phones. How do i interface my POTS phones with this; can i just crimp an > RJ45 connection on the end of the phone cord? either that or just plug an rj11 into the rj45 (it still mates on the center pair either way) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 255
Spencer, We have a web interface NV*GUI that does just this. We are building it for another company and they are allowing us to release it GPL. Out very soon...maybe this weekend. Justin Newman Newman Telecom, Inc. -- Message: 3 Date: Thu, 28 Apr 2005 20:41:09 +0100 (BST) From: "G.Marshall" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Web interface Suggestions To: "Asterisk Users Mailing List - Non-Commercial Discussion" Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain;charset=iso-8859-15 > Has anyone come across any software that can control adding/editing > SIP extension properties and perhaps dial plan properties on a context > basis. What I mean is I would like it so an admin user from Company A > can manipulate > properties for extensions in his context but not in another Companies. I > know AMP does something similar > to this but from what I understand it does not allow for different users > at different companies to control > only things that pertain to them. In my spare time, I am developing a php webfrontend to realtime asterisk database which modifies dialplan, users etc. Should not be too difficult to add a login facility which means the user can see their own context only. Regards, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: NVBackgroundDetect
> Date: Tue, 03 May 2005 23:14:18 -0600 > From: Joseph <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] NVBackgroundDetect > > Is anybody using NVBackgroundDetect to detect fax signal on SIP protocol > from ATA? > > -- > #Joseph We're using NVBackgroundDetect with SIP and IAX. Several of our products/customers use it successfully -- and in volume. In fact, I think we've only experienced one or two missed detections, which was most likely configuration. For SIP, we've tested all the Sipura (SPA-1000, SPA-1001, SPA-2000, SPA-2100, and SPA-3000), Mediatrix, D-Link, and Linksys adapters. We've tested several providers and * to * configs. No problems. Make sure you follow the wiki instructions for setup and that you understand usage. Link quality, protocol, provider, and mo/do determine that actual fax send/receive results (beyond detection). Great results here as well on send/receive, except on the software side with PDF and TIFF GPL libraries. T.38 and others attempt to solve FOIP, but G711 is working well for most of our clients. The modules have been tested over LAN, DSL, cable, T1, and OC3. DSL and cable worked well, except in some areas cable had problems. In some situations, DSL performance was worse, but generally it was better. BTW- On another note, Cisco recently purchased Sipura. Hopefully they won't kill the company like they have others. Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of FAX
Joseph, Are you still having problems with NVFaxDetect and NVBackgroundDetect on Gentoo? Give me a call or e-mail me. It works on nearly every distro we've evaluated. Justin Newman [EMAIL PROTECTED] Newman Telecom, Inc. > Date: Wed, 11 May 2005 15:15:01 -0600 > From: Joseph <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Status of FAX > > On Wed, 2005-05-11 at 14:18 -0600, [EMAIL PROTECTED] wrote: > > Hi people, what is the current status of send/receive fax on asterisk > > extensions, i dont want to receive the fax and send an email orviceversa, i > > want to connect a standard fax machine to a Linksys' ATA (FXS RJ11 port) . > > Webdoc?, pointers? > > Thanks > > I would like to do the same (not interested with fax to email), just > would like to receive and send a fax on an extension using hylafax or > standard fax. > > There is a perfect solution called NVBackgroundDetect application (that > you need to compile yourself); as I don't know when will it appear as a > standard feature on stable version. > > I've tried on Gentoo but didn't have much luck. > > -- > #Joseph > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 117
> Date: Sun, 15 May 2005 15:17:53 -0700 > From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] 911 Options > To: Ira Burton <[EMAIL PROTECTED]>, Asterisk Users Mailing List - > > On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote: > > I am curious if anybody has pointers on the best way to get the 7 > > digit PSAP number for an area. I am thinking about making a '911' > > extension that will dial the PSAP number, wait for the PSAP to answer > > and play a message giving the address of the originating call, and > > replay the the information every three minutes. I am concerned what > > may happen if my children try to dial 911 in an emergency but do not > > yet know our address. > > > > You can buy them on CD, however to do E911 you have to have a special > trunk to the switch that the PSAP is off of, which transmits the E parts > of E911 not just the audio. > > Where to buy them I dont know offhand, I do specifically recall seeing > pages that sold national CDs (how adt, onstar, even other PSAPs contact > a specific PSAP when needed). You can buy them from a few different vendors. Last I checked it was like $50,000. You can also just call up your state administrators for PSAP. Keep in mind that numbers can change, though, which is why updates are important. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Asterisk Fax
> Date: Tue, 17 May 2005 13:08:14 -0600 > From: Rich Adamson <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Asterisk Fax > > > I have read a lot about the thread of faxing support in Asterisk as > > well as spanDSP. However, either I don't fully understand other > > people's applications or may be what I'm trying to do is different > > from what others are trying to do. > > > > I have a very simple setup. I have an asterisk server with a TE110P > > connected to the PSTN via T1 PRI (Asterisk A). I have another > > asterisk server with a TDM40B board with 4 FXS ports (Asterisk B) > > where I have an actual physical fax machine connected to one of these > > ports. These two machines are in two separate locations connected via > > a point-to-point T1 circuit. What I wish to do is program a DID on > > the T1 so that when the call comes into Asterisk A on that DID, it > > will be routed via IAX2 to Asterisk B, which will in turn patch the > > call through the FXS port where the fax is connected. Obviously, I > > also wish to be able to send faxes in a similar opposite direction. > > > > Is this possible with Asterisk? Will it work? Search for NVFaxDetect and NVBackgroundDetect for detection. We use fax over IAX with 98% success rate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] NVFaxDetect on Gentoo
> > Date: Fri, 20 May 2005 13:33:54 -0600 > > From: Joseph <[EMAIL PROTECTED]> > > Subject: Re: [Asterisk-Users] NVFaxDetect on Gentoo > > > > On Fri, 2005-05-20 at 12:06 -0300, Juan Luis Moyano wrote: > > > Hi, I've merged asterisk-0.9.0 on a gentoo (kernel 2.4) system using > > > portage ebuilds. I've just got NVFaxDetect .c files from Justin Newman > > > and I'm about to install them. I want to know which is the best way to > > > accomplish this. Thanks in advance. > > > > I would like to know that too. > > I've initiated a discussion through Gentoo Bugzilla see: > > http://bugs.gentoo.org/show_bug.cgi?id=92747 > > > > If only Mr. Newman gave us the link to the source code we could write a > > ebuild to install/compile it as an addition. > > > > -- > > #Joseph Joseph, We are setting up a location for all the GPL modules. If you have any specific requirements, please let us know. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy
> Date: Thu, 23 Jun 2005 08:50:50 +0200 > From: "Robert Rozman" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * > - Euroisdn Italy > > I'm pulling my hair down and getting bold :-) . I have Asterisk between > Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff > Asterisk) Plenty of experience with the Panasonics, but not the EuroISDN. Contact me offline if you have KXTD816 questions. Regards, Justin [EMAIL PROTECTED] Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for Honduras DIDs, Origination, Termination
Looking for Honduras DIDs, origination, and termination. Contact me offlist. Regards, Justin Newman [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Software SIP fax client
We have something on the way. Regards, Justin Newman [EMAIL PROTECTED] > - Original Message - > > > Does anyone know of a software SIP fax client? Something I can install = > > on a PC which connects to the asterisk server and sends/receives faxes? = > > Something like XLite - but to fax instead of to phone. > > =A0 > > I know of the "fax machine connected to an ATA" solution, but that's not = > > really what I'm looking for :-) > > =A0 > > Thanks > > -Manuel > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming Fax Service question
If you need to dial additional digits after pickup, use the D(...) command with Dial. Why not just send the call to another extension or DID? To detect fax on the line, you can use NVFaxDetect or NVBackgroundDetect. More information on the Tikiwiki. http://www.voip-info.org/tiki-index.php?page=NVFaxDetect http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect Justin -- Justin J. Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NVFaxDetect errors on make
Look in app_nv_backgrounddetect.c and app_nv_faxdetect.c. Near the top you should see: // Use the second one for recent Asterisk releases #define CALLERID_FIELD cid.cid_num //#define CALLERID_FIELD callerid Change it to: // Use the second one for recent Asterisk releases //#define CALLERID_FIELD cid.cid_num #define CALLERID_FIELD callerid Save, recompile, and you should be set. -- Justin Newman Newman Telecom, Inc. [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Print-to-Fax client
We have a client that's almost done - it allows faxing from Windows/Linux via e-mail or SIP. Your Asterisk server can decide what to do with it at that point... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax receive issues and NVFaxDetect
> From: "Chris Tuska" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Fax receive issues and NVFaxDetect
>
> [macro-faxreceive]
> exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
> exten => s,7,rxfax(${FAXFILE})
> exten => s,103,SetVar([EMAIL PROTECTED])
> exten => s,104,Goto(7)
>
> [from-Sipmedia2]
> ;second line in or Fax line
> exten => s,1,Answer
> exten => fax,2,Goto(fax,2901,1)
>
> [fax]
> exten => 2901,1,Macro(faxreceive)
>
> exten => h,1,System(/var/lib/asterisk/scripts/mailfax "${FAXFILE}"
"${EMAILADDR}" "${CALLERIDNUM}" "${CALLERIDNAME}")
I'm a little bit confused here. Assuming the "from-Sipmedia2" context is
handling your SIP appliance, how about the following (just moved around a
little of yours and added a line)?
> [macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,rxfax(${FAXFILE})
exten => s,3,System(/var/lib/asterisk/scripts/mailfax "${FAXFILE}"
"${EMAILADDR}" "${CALLERIDNUM}" "${CALLERIDNAME}")
[from-Sipmedia2]
exten => s,1,Answer
; You need this line in here
exten => s,2,NVFaxDetect
exten => fax,1,Goto(fax,2901,1)
[fax]
exten => 2901,1,Macro(faxreceive)
Jusitn Newman
[EMAIL PROTECTED]
___
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[Asterisk-Users] Zap Detect called party pickup
>Date: Fri, 25 Mar 2005 18:39:26 +0100 (CET) >From: Peter Svensson <[EMAIL PROTECTED]> >Subject: Re: [Asterisk-Users] Zap Detect called party pickup > >On Fri, 25 Mar 2005 [EMAIL PROTECTED] wrote: > >> I have been playing with getting the sample.call file to work by >dropping it into >> /var/spool/asterisk/outgoing. The process works to the point of calling >the desired >> number and plays the message. The problem is that the message starts >playing almost >> immediately, so if the called person takes 2 or 3 rings to pick up the phone, half the >> message has already been played. > >You need answer supervision on your line. It is available on isdn lines >and some analogue lines. We have a module that will do this detection. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Using call.sample on Zap hardware - Answering problem
> From: "Patrick Healy" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Using call.sample on Zap hardware - > Answering problem > I've got a X100P connected to a POTS line and am using it to call out to > play a recorded message. I drop a copy of sample.call into > /var/spool/asterisk/outgoing and Asterisk picks up the line and initiates > the call. The problem is that the recorded message starts immediately and > doesn't wait for the called party to pick up the phone. When I try this > same process with a SIP extension, the process works like a champ, it just > fails on the Zap interface. Check out NVLineDetect - it detects answer, busy, ringing, and other states. It will be available soon. Read more on http://www.voip-info.org or e-mail [EMAIL PROTECTED] for the code. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Fax receive issues and NVFaxDetect
>> I removed everything but what I needed to get the fax and email it to
>> myself. So this is all I have, Thanks..
>Well, then you need to renumber the priorities! Below is a start. Your
>code does not do any emailing, by the way... what happens when you set
>the EMAILADDR variable? Nothing... you need to add a handler script to
>send the email.
>
>[macro-faxreceive]
>exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
>exten => s,2,rxfax(${FAXFILE})
>exten => s,3,SetVar([EMAIL PROTECTED])
>exten => s,3,System(SOME EMAILING CODE HERE)
>exten => s,5,Goto(1)
>
>--Luki
Or you can use NVFaxEmail to eliminate the scripting, however it still
requires SpanDSP and RxFax.
Look on the wiki for more information. It's either NVFaxEmail or
NVEmailFax - can't remember right now.
[context]
exten => s,1,NVFaxDetect
exten => fax,1,NVFaxEmail([EMAIL PROTECTED])
Justin Newman
Newman Telecom, Inc.
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[Asterisk-Users] re: Problem: Compiling error for SpanDSP
> Date: Tue, 29 Mar 2005 21:43:06 -0500 > From: "KMZ Enterprises" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Problem: Compiling error for SpanDSP > app_rxfax > > After resolving my earlier problem in updating the apps Makefile with the > patch for SpanDSP, I encountered another problem when I executed the > "Make" > utility from /usr/src/asterisk. I obtained an error as shown below. Not > sure on how to resolve the problem. > > > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes > -Wmissing-declarations -g -Iinclude -I../incl > ude -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS > -DASTERISK_VERSION=\"CVS-HEAD- > 03/29/05-21:11:20\" -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\" > -DASTETCDIR=\"/etc/asterisk\" -D > ASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" > -DASTVARRUNDIR=\"/var/run/asterisk\ > " -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" > -DASTCONFPATH=\"/etc/asterisk/ > asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" > -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" > -DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o app_rxfax.o > app_rxfax.c > app_rxfax.c: In function `phase_e_handler': > app_rxfax.c:86: structure has no member named `callerid' > make[1]: *** [app_rxfax.o] Error 1 > make[1]: Leaving directory `/usr/src/asterisk/apps' > make: *** [subdirs] Error 1 > > Regards, > Kerry For a while, the structure switched over. If you are getting an error with "callerid" in your build, try searching the code (app_rxfax.c) and replace the occurances with "cid.cid_num". Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Fax detection
> Date: Wed, 30 Mar 2005 14:29:06 -0500
> From: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Fax detection
>
> Hello,
>
> I'm attempting to configure my office Asterisk server to do fax
> detection for each one of our DID's configured for different users.
> Each person in our office has their own phone number, and I want each
> to do both voice & fax.
>
> Fax detection works great when configured like this:
>
> exten => fax,1,Macro(faxreceive)
> exten => fax,2,SetVar([EMAIL PROTECTED])
> exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
> "${CALLERIDNUM} ${CALLERIDNAME}")
>
> This will detect any incoming fax for this particular context and
> basically work as expected.
> I attempted to do some callerid matching to ensure the correct person
> gets their fax.
> When configured like this, Asterisk doesn't match the incoming DID to
> the fax user in question.
>
> exten => fax/3172152560,1,Macro(faxreceive)
> exten => fax/3172152560,2,SetVar([EMAIL PROTECTED])
> exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
> "${CALLERIDNUM} ${CALLERIDNAME}")
>
> This didn't work!
> I then tried this:
>
>
> exten => fax,1,Macro(faxreceive)
> exten => fax/3172152560,2,SetVar([EMAIL PROTECTED])
> exten => fax/3172152561,2,SetVar([EMAIL PROTECTED])
> exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
> "${CALLERIDNUM} ${CALLERIDNAME}")
>
> this also didn't work, although it did everything but set the e-mail
> variable.
> Any ideas?
> Niles
Send your output from Asterisk. Also, your second example needs to be
reordered (priorities are not numbered right).
Are you coming in on ZAP?
Justin Newman
Newman Telecom, Inc.
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[Asterisk-Users] Re: "Decent" sub-$100 SIP phone.
Too bad this wasn't a couple weeks ago. We just sold a huge lot of unused Polycom IP300's for $99/each. --- Date: Mon, 09 Jan 2006 15:28:28 -0500 From: Ken D'Ambrosio <[EMAIL PROTECTED]> Subject: [Asterisk-Users] "Decent" sub-$100 SIP phone. To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy environment (a cleanroom), the less overall static, the better. Someone suggested the Polycom 301's, but I'd lose money on them. [I'll go with them if I have to, as I'm making money elswhere, but still...] So, does anyone have any suggestions for decent sub-$100, professional-looking SIP phones? Thanks! Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Liquidation: Cisco; Polycom; D-Link; MediaTrix, Colubris - Highly Reduced Prices
We have extra equipment that was over-ordered or unused. All of the equipment is brand new. The equipment has been highly discounted to move quickly - the last set of equipment sold in 48 hours. If this equipment is of interest to you, call or e-mail quickly. Buy on VOXILLA and SAVE $300 each (Cisco routers & switches): http://store.voxilla.com/customer/home.php?cat=259 For Sale (all new): 4 Cisco 3550 24 PT Switch POE$2265 5 Cisco 1841 Router & WIC $1608 2 D-Link DVG-1120S ATA (SIP)$169 5 MediaTrix 2102 ATA (SIP) $109 1 Polycom SoundStation Premier E$499 9 Colubris CN3200 AP w/ 100 UAC $715 Special: Cisco 3550 is $1965 on http://store.voxilla.com Special: Cisco 1841 is $1308 on http://store.voxilla.com Feel free to make an offer in whole or in part. E-mail [EMAIL PROTECTED], [EMAIL PROTECTED], or call (503) 704-9151. -- Justin J. Newman Chief Executive Officer Newman Telecom, Inc. "Ok... A CEO who will actually field personal phone calls and this level of support... that rates first class to me..." -- Daniel Creed, Jefferson Wells Sales (800) 653-3270 Ph/Fax (503) 914-5181 Mobile (503) 704-9151 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Remote call pick-up
Does Asterisk have remote call pickup (i.e. reverse transfer)? We wrote a module for customer which does this. Justin - Date: Wed, 05 Oct 2005 06:54:57 +1300 From: Damian Funnell <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Remote call pick-up Hi, Does anyone have remote call pick-up working on * (either via SIP or otherwise)? If so then can you post your features.conf, sip.conf and/or zapata.conf? We can't seem to get this (seemingly simple) function to work. Cheers, Damian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looks like Nufone is changing around...
Looks like Nufone is making some positive changes... --- NuFone Announces the Creation of a New Executive Management, Support Team NuFone Inc., a premium-service provider specializing in hosted SIP and IAX VoIP solutions, is proud to announce the creation of a new executive management and support team. Eugene, OR (PRWEB) September 2, 2006 -- NuFone, the world's first commercial provider of IAX-based VoIP services, announced today the creation of a new management and support team to further solidify its dedication to providing reliable VoIP solutions to carrier, enterprise and residential environments. Composed of 5 executive team members, who are highly experienced in the areas of business, sales and support, will provide the necessary leadership to properly manage NuFone. "In the past, NuFone always had trouble properly managing and supporting our customer's needs. It has always been my goal to form a proper team to deliver the support our customers demand," Jeremy McNamara, founder and CTO of NuFone, said. "By listening to our customers, we were able to determine our weaknesses and have formed a proper team to bring NuFone to the next level." The following are the executive members of the NuFone management and support team: Allan Noorda, President and CEO. Noorda has been engulfed in the advancement of the Telecommunications Industry for over the past 10 years. Noorda recently comes from Newman Telecom, where he was the VP of Sales and Marketing. He brings energy and an understanding of customer needs as well as technology to direct the daily operations of NuFone. Jeremy McNamara, Founder and CTO. Over the past 10 years, McNamara has assisted in the development and deployment of several ISPs, ITSPs and Application Service Providers around the United States. McNamara also has extensive development, testing and deployment expertise with Asterisk PBX-based solutions. Greg Merriweather, Support Specialist. Merriweather has been providing operating system, hardware and application support for the past 10 years including working as a support engineer for Ford and Global-Crossing before assisting in the operation of NuFone beginning in early 2003. Leon Salisbury, Senior Engineer. Salisbury has over 20 years of programming and engineering experience with a wide assortment of programming languages including Assembler, Perl, HTML, C and hardware including PICs, 68xx Series, various DSP and embedded x86 platforms. Krystina Patterson, Customer Relations. Patterson has been working with the public for the past 4 years in marketing and customer relations. Patterson is well versed in problem solving and determining customer needs. About NuFone NuFone was originally deployed by Jeremy McNamara in January 2002, as an IAX-based solution for Asterisk PBX based users. NuFone has since grown into a leading provider of SIP and IAX based VoIP solutions for thousands of customers in the United states and more than 60 countries world-wide. NuFone is a privately held corporation based in Eugene, OR. ### ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Nufone making changes
Justin Newman wrote: Looks like Nufone is making some positive changes... Thanks Justin, but Asterisk-users is for Asterisk discussion only. Perhaps a more appropriate list would be asterisk-biz. Jeremy McNamara Noted. Nonetheless, looks like you guys are making progress. :p ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: FAX handling
Let me know if you guys need help with this... Justin -- Message: 15 Date: Mon, 4 Sep 2006 17:16:00 -0400 From: "Technical Support" <[EMAIL PROTECTED]> Subject: RE: [asterisk-users] FAX handling To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Look into NVDETECT, and fax2mail script on www.generationd.com Fax detection is automatic MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Modifying Voicemail App
You will need to modify /usr/src/asterisk/apps/app_voicemail.c. Fairly easy task. On another note, I'm surprised the IVR within apps such as voicemail isn't drawn out into a app specific app/dialplan. The application flow could then be easily customized by end users. This wouldn't be too hard to do... -J > Date: Sun, 16 Oct 2005 22:57:48 -0700 (PDT) > From: Neil Skowronek <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Modifying Voicemail App > > I want to add things to the prompts like: > > "mark urgent" > > "add to message" > > "pause while recording message" > > Any examples of how to do this? > > I'd also like to switch around prompts, not simply > edit the sound files. > > Is it an agi, special dailplan, patching the > app_voicemail.c file? All three? > > Any input/examples are welcome. > > -thanks > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can't compile ast_*fax
Try adding "#define _GNU_SOURCE" before including lock.h and pthread.h. Justin > Date: Mon, 17 Oct 2005 14:29:12 -0400 > From: "Carlos Alperin" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Can't compile ast_*fax > > I tried to compile app_rxfax.c & app_txfax.c, and always fails. > > > > Even if I put this on a different directory and I do gcc -pipe app_rxfax.c I > get the same > > > > /usr/include/asterisk/lock.h: In function `ast_mutex_init': > > /usr/include/asterisk/lock.h:300: `PTHREAD_MUTEX_RECURSIVE' undeclared > (first use in this function) > > /usr/include/asterisk/lock.h:300: (Each undeclared identifier is reported > only once > > /usr/include/asterisk/lock.h:300: for each function it appears in.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CDMA phone line for Asterisk?
It may be easier to buy a CDMA card or module made for Asterisk. There are several GSM products and at least one company that has CDMA products. Let me know if you need more information. -J > Date: Tue, 18 Oct 2005 01:49:16 +0600 > From: Widyachacra Rajapaksha <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] CDMA phone line for Asterisk? > > Dear friends, > > im very new to asterisk, even diz z my 1st mail to the list. am working a > small company & it has two main CDMA telepone connections. now they wants to > deploy a pbx & get out 20 nods(telephone extension lines). so is it possible > with Asterisk? & our both CDMA phones are HUAWEI ETS2000 Series models, > those came with USB serial converter data cable(diz can connect to de pc > USB+work with linux ti_usb_3410_5052 kernel module). > > so please guide me to do diz project... > > you can find de HUAWEI CDMA phone details > http://www.huawei.com/mobileweb/en/products/view.do?id=152 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ACD calls to busy agents
> Date: Mon, 17 Oct 2005 14:20:18 -0500 > From: Corey Frang <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] ACD calls to busy agents > > So, I'm looking into using PauseQueueMember and unpause queuemember > > How the heck to you get Unpause to run, no matter what, after the call > is over? > > The "g" argument to Dial only works when the >called< party hangs up. Isn't this when you want to Unpause? If the called was still on the line, wouldn't the agent be busy? If you are thinking that the agent's channel will still be active after the call, just place the Unpause before you release the channel (hangup). Otherwise, with the first senario: exten => s,1,PauseQueueMember(...) exten => s,2,Dial(...,g) exten => s,3,UnPauseQueueMember(...) You can also add, to trap most conditions: exten => s,103,UnPauseQueueMember(...) ; Busy exten => T,1,UnPauseQueueMember(...) ; Absolute Timeout exten => t,1,UnPauseQueueMember(...) ; Timeout exten => i,1,UnPauseQueueMember(...) ; Invalid exten => h,1,UnPauseQueueMember(...) ; Hangup > Using the "h" extension appears to be doing nothing... > > Is there any way we could add a feature to the "pausequeuemember" that > basically says "As long as this channel is open, this member is paused" > so that way when they hang up they are unpaused automatically? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problem with compiling spandsp
Use pre10 or pre11. > Date: Mon, 17 Oct 2005 12:53:43 -0700 > From: "Administrator" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Problem with compiling spandsp > > Actually I am using 0.0.2pre21, also tried pre20finally got a > different error after trying just about everything including deleting > the source dir and unpacking again, editing makefile again, etc. > > app_rxfax.c: In function `rxfax_exec': > app_rxfax.c:265: error: structure has no member named `logging' > app_rxfax.c: At top level: > app_rxfax.c:61: warning: 't30_flush' defined but not used > make[1]: *** [app_rxfax.o] Error 1 > make[1]: Leaving directory `/usr/src/asterisk-1.0.9/apps' > make: *** [subdirs] Error 1 > > Maybe I'm not editing the makefile correctly? I am cutting/pasting from > the patchfile so I know it's not a typo. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Fax2Mail
I don't know of a comprehensive guide, but you can set it up using
NVFaxDetect, NVFaxEmail, and SpanDSP or Hylafax. NVFaxEmail can pull the
e-mail addresses from it's own config, voicemail.conf, a database, or thru
realtime.
Simple extensions.conf:
[incoming-dids]
exten => _541359,1,NVFaxDetect(...) ; Make sure this is a fax
exten => fax,1,NVFaxEmail(...,${CALLERID},pu,...) ; Receive and e-mail PDF
with user lookup
-J
> --
>
> Message: 10
> Date: Tue, 18 Oct 2005 07:39:10 -0700 (PDT)
> From: David <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Fax2Mail
>
> Hello,
>
> Is there or can anyone provide a comprehensive guide (designed for
Linux/Asterisk novices) to installing/setting up Asterisk in order to
support Fax2Mail service?
>
> In my case, I would like Asterisk to receive fax calls to predefined
numbers (ranges) and to associate each of these numbers to email addresses.
>
> Thank you in advance.
>
> David
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[Asterisk-Users] Re: merchant account
What about SMS or LEC billing? The second may be more difficult, but there are many solutions with SMS, even if it is just transferring contact information (instead of hiring operators to typing in data with the keypad). The first can be used alone or in conjunction with credit cards and direct widthdrawals. 900 numbers work well, but they are expensive. Probably the 2nd easiest solution, aside from credit cards (if you can tackle the data collection piece). Credit cards are difficult because of security problems, plus the market you are targeting may not have legal access to these cards. :) That means chargebacks. Direct withdrawal (ACH/EFT) can be even more problematic than credit cards, although it is inexpensive. > Date: Fri, 21 Oct 2005 00:03:24 -0400 (EDT) > From: "Jon Pounder" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] merchant account > > > On Thu, 2005-10-20 at 23:07 -0400, Jon Pounder wrote: > >> my suggestion is to issue your own "cards" even if they don't physically > > > > not an option, the customer that is needed this is setting up an adult > > service and he believes that since its just for adult entertainment that > > no one will front the money to have a prepaid account like that. He > > doesnt trust the girls so he doesnt want them to have access to any of > > the billing information. > > > > While not all merchant accounts support adult services, enough do that > > if anyone said they had one that worked with phone only interfaces I > > would investigate, its starting to look like the person will have to > > signup online and all that. Which isnt a bad thing since it does > > involve some online aspects in addition to voice. > > another option is 900 service, then you don't have to worry about > creditcards at all. > > iBill used to have something called web900 where you phoned and got a > generic token to use on the web, you could just accept those tokens (just > a number) and not deal with the 900 part at all. Basically they issue you > a file with tokens and you just erase them as they get redeemed, when you > need more, you get a new file from them. (iBill is adult friendly btw, > porn is their bread and butter) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: faxdetect on voicemail
We have a voicemail solution, but here's a way you can do it in your
dialplan below. This also allows you to check messages remotely and/or
disable voicemail. Disabling voicemail is desirable on real fax machine or
credit card lines.
[macro-resvoice-handlevoicemail]
exten => s,1,DBGet(voicemail=VOICEMAIL/${ARG1})
exten => s,2,GotoIf($["${voicemail}" = "NO"]?done,1:3)
exten => s,3,Answer
exten => s,4,Wait(1)
exten =>
s,5,NVBackgroundDetect(/var/spool/asterisk/voicemail/default/${ARG1}/unavail
,t)
exten => s,6,Voicemail(s${ARG2})
exten => s,7,Hangup
exten => fax,1,NVGetVoicemailUser(${ARG1})
exten => fax,2,GotoIf($["${VMB_USER_EMAIL}" = ""]?s,6:3)
exten =>
fax,3,NVFaxEmail([EMAIL PROTECTED],${VMB_USER_EMAIL},p,You,${VMB_USER_
FULLNAME})
exten => fax,4,Hangup
exten => #,1,VoicemailMain(${ARG2})
exten => #,2,Hangup
exten => done,1,NoOp
Justin Newman
>Date: Thu, 27 Oct 2005 09:59:30 +0330
>From: Paradise Dove <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] faxdetect on voicemail
>
>hi,
>is there anyway to just enable faxdetection in voicemail?
>
>thanks,
>paradise dove
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[Asterisk-Users] Re: Fax between Asterisk SIP clients
SpanDSP comes with app_rxfax and app_txfax. If you need to send a fax, you can initiate this from one of your ATA connected fax machines. For automation, you can use call files (a sample is at /usr/src/asterisk/sample.call) or the Asterisk Manager API. Another method, although less automated- you could have an extension that you dial into to kick the whole proces off. If you need fax detection, you can try NVFaxDetect and NVBackgroundDetect. For PDF/TIFF fax to e-mail, try NVFaxEmail. Contact me off list if you need more help. Justin > Date: Wed, 2 Nov 2005 10:53:29 -0800 > From: Andy Kuo <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Fax between Asterisk SIP clients > > Hi all, > I'm looking for a fax solution with Asterisk. I would like the users to be > able to hook up regular fax machines to their SIP ATA's and send/receive fax > from PSTN and/or other SIP clients. > My goal is: > fax machines <-> SIP ATA - Asterisk - T1(TE406E) <-> fax on > PSTN > It looks like Hylafax will allow me to receive fax from PSTN, but not send > to PSTN. I also tried Spandsp, and it seems to receive fax ok from ATA's, > but I can't figure out how to have it automatically forward the fax file to > fax machines on PSTN or other SIP extensions. > Can I have Spandsp dial and send the fax to the destination automatically? > Are there other software / hardware solutions that can help me achieve my > goal? > Please advise. > Thanks to any help/ideas. > AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Blind transfer from queue into another queue
> I want to transfer a call that has come into one queue, and that I have > already accepted, into another queue. > > When I try this asterisk tells me "Transfer attempted with no > appropriate bridged calls to transfer". > It is possible to forward the call to another person, but forwarding > into a queue fails. > Is forwarding from one queue into another possible at all? Just transfer to another extension that points to a queue. You can also try NVQueue. Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clearwire and Asterisk
Has anyone had problems using Clearwire, VOIP, and/or Asterisk? Just curious... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NVFaxDetect and NVBackgroundDetect on Asterisk 1.2
If you are unable to build NVFaxDetect and/or NVBackgroundDetect on Asterisk 1.2 (and/or AMP or @home Beta), make the following changes: 1) Above the following line near the top, in both files: #include Add: #include 2) In NVBackgroundDetect, to get rid of the trigraph warning, search for "??)" and replace it with "?)". 3) Rebuild Asterisk from /usr/src/asterisk with "make && make install". 4) Restart Asterisk with "restart now" from the CLI. The new release will have this modification. Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Nufone Echo Test
Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is NOT dead. It is working and I just added more funds into my account. You may also consider Asterlink. I'm a new client there, their support is a little slow, sometimes irresponsive (you need to send several messages until they notice you), they also have a misconfigured mail server but other than these problems, so far so good. Does anyone have new information on their echo test? The one that was posted some time ago seems to no longer work. Dial(IAX2/[EMAIL PROTECTED]) John Novack Did echo disappear? Justin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Desired apps
Is there a list of apps or "desired features" for users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Broken Call Screening
You need to modify app_queue.c to hold off on bridging until the receiving party has accepted the call. If the receiving party rejects (hangup, digit other than '1', timeout, etc), leave or put the calling party back in at close to the same level.Justin--Date: Tue, 14 Nov 2006 10:14:04 -0500From: "Gary T. Giesen" <[EMAIL PROTECTED]>Subject: [asterisk-users] Broken Call Screening...I have a cell phone added to a queue as a local extension (member =>Local/299). I want the cell phone to be able to reject calls to thequeue without the person sitting in the queue being hung up on, etc.The way my dialplan is set up, the person hits 1 to answer the calland any other key to reject it. It works flawlessly in that regard.If it goes to the cell phone voicemail, it works great too, it timesout and rejects the call, all without the caller knowing. Where itbreaks is when the person answers the cell phone and then hangs upwithout any input or letting it time out. The music on hold is stoppedand the caller is left there with dead air. Does anyone have any ideason how to fix this or a better way to implement this?Output when the call is dropped: -- Channel 0/3, span 1 got hangup request -- User disconnected -- Stopped music on hold on Local/[EMAIL PROTECTED],2Nov 13 16:21:26 WARNING[12709]: res_features.c:1374 ast_bridge_call:Bridge failed on channels Local/[EMAIL PROTECTED],2 and Zap/3-1 -- Hungup 'Zap/3-1' -- Local/[EMAIL PROTECTED],1 answered SIP/7960A-Gary1-63f2 -- Stopped music on hold on SIP/7960A-Gary1-63f2...Regards,Gary___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
