Re: [asterisk-users] Voicemail Transcription with openai/whisper
I really love this idea. Thanks for sharing. I've been looking for a good way to provide this service to my customers. Hopefully this will work for me too. Thanks, Dave On 11/27/2022 8:08 AM, Doug Lytle wrote: Everybody, I've recently discovered openai/whisper and have been trying in earnest to get this working with Asterisk for voicemail transcriptions (Currently using the NerdVittles script with IBM Watson) https://github.com/openai/whisper After spending several hours today, I've successfully integrated my home Asterisk 16 voicemail with Whisper. Once I have followed the instructions for setting up an API server https://blog.deepgram.com/how-to-build-an-openai-whisper-api/ Initially, I setup a quad core VM to test this with, but discovered that without a dedicated card for the inference that it was horribly slow. So, I've set up testing on my desktop (Kubuntu 20) since I have an nVidia GTX 1060 installed. For the integration with Asterisk, I'm using a slightly modified script from nerdvittles IBM Watson script sendmailibm That can be found on their website https://nerdvittles.com/free-asterisk-voicemail-transcription-with-ibms-stt-engine/ I will probably find a low cost nVidia video card and get a stand alone Linux box running to handle this project. If you're interested in the details, let me know. Doug -- _ -- 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] Voicemail Transcription with openai/whisper
On 11/27/22 09:22, Greg Troxel wrote: Thanks for posting. As I'm running asterisk on a PC Engines apu2, I don't need the details as it is obviously unworkable, but it's great to see non-cloud progress. Greg, Just a note, This would work if you have the API server running on a Linux x86 box. Then Asterisk would be using curl and python to communicate with that API Linux box. Doug-- _ -- 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
[asterisk-users] Voicemail Transcription with openai/whisper
Everybody, I've recently discovered openai/whisper and have been trying in earnest to get this working with Asterisk for voicemail transcriptions (Currently using the NerdVittles script with IBM Watson) https://github.com/openai/whisper After spending several hours today, I've successfully integrated my home Asterisk 16 voicemail with Whisper. Once I have followed the instructions for setting up an API server https://blog.deepgram.com/how-to-build-an-openai-whisper-api/ Initially, I setup a quad core VM to test this with, but discovered that without a dedicated card for the inference that it was horribly slow. So, I've set up testing on my desktop (Kubuntu 20) since I have an nVidia GTX 1060 installed. For the integration with Asterisk, I'm using a slightly modified script from nerdvittles IBM Watson script sendmailibm That can be found on their website https://nerdvittles.com/free-asterisk-voicemail-transcription-with-ibms-stt-engine/ I will probably find a low cost nVidia video card and get a stand alone Linux box running to handle this project. If you're interested in the details, let me know. Doug -- _ -- 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] VoiceMail() stops dialplan processing
On Monday 03 October 2022 at 14:14:54, Joshua C. Colp wrote: > On Mon, Oct 3, 2022 at 9:11 AM Antony Stone < > > antony.st...@asterisk.open.source.it> wrote: > > Hi. > > > > I have a dialplan which calls the VoiceMail() application, and I'm > > getting the following behaviour: > > - if the inbound caller leaves a message, then presses #, and then > > presses 1 to accept the recording, everything works as expected and the > > dialplan continues processing after the line containing VoiceMail() > > > > - if the inbound caller leaves a message and then hangs up, the diaplan > > simply stops executing with a message such as: > > > > [2022-10-03 13:02:23.355976] pbx VERBOSE[19022][C-0556]: pbx.c:4413 > > in __ast_pbx_run: Spawn extension (RecordVM, 00xx74xx88xx90, 2) exited > > non-zero on 'SIP/TrunkOne-0c12' > > > > The subsequent commands in the dialplan do not get processed. > > This is fundamentally how dialplan works. If a channel hangs up, then > normal dialplan execution stops. I suppose that fits other situtations, yes. > > Can anyone suggest either why this would happen and how to get the > > dialplan to continue processing under all circumstances, or at least how > > to investigate futher what is causing this to happen? > > > > I'm sure that leaving a message and hanging up the call should be valid > > because that's what the default greeting message tells people they can > > do. > > It is. If you're needing to do something afterwards, then the 'h' extension > or hangup handlers are used to execute logic when the channel is hung up. Okay, sounds simple enough - thanks, Antony. -- RTFM may be the appropriate reply, but please specify exactly which FM to R. Please reply to the list; please *don't* CC me. -- _ -- 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] VoiceMail() stop dialplan processing
On Mon, Oct 3, 2022 at 9:11 AM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > Hi. > > I have a dialplan which calls the VoiceMail() application, and I'm getting > the > following behaviour: > > - if the inbound caller leaves a message, then presses #, and then > presses 1 > to accept the recording, everything works as expected and the dialplan > continues processing after the line containing VoiceMail() > > - if the inbound caller leaves a message and then hangs up, the diaplan > simply stops executing with a message such as: > > [2022-10-03 13:02:23.355976] pbx VERBOSE[19022][C-0556]: pbx.c:4413 in > __ast_pbx_run: Spawn extension (RecordVM, 00xx74xx88xx90, 2) exited > non-zero > on 'SIP/TrunkOne-0c12' > > The subsequent commands in the dialplan do not get processed. > This is fundamentally how dialplan works. If a channel hangs up, then normal dialplan execution stops. > > > Can anyone suggest either why this would happen and how to get the > dialplan to > continue processing under all circumstances, or at least how to > investigate > futher what is causing this to happen? > > I'm sure that leaving a message and hanging up the call should be valid > because that's what the default greeting message tells people they can do. > It is. If you're needing to do something afterwards, then the 'h' extension or hangup handlers are used to execute logic when the channel is hung up. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- 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
[asterisk-users] VoiceMail() stop dialplan processing
Hi. I have a dialplan which calls the VoiceMail() application, and I'm getting the following behaviour: - if the inbound caller leaves a message, then presses #, and then presses 1 to accept the recording, everything works as expected and the dialplan continues processing after the line containing VoiceMail() - if the inbound caller leaves a message and then hangs up, the diaplan simply stops executing with a message such as: [2022-10-03 13:02:23.355976] pbx VERBOSE[19022][C-0556]: pbx.c:4413 in __ast_pbx_run: Spawn extension (RecordVM, 00xx74xx88xx90, 2) exited non-zero on 'SIP/TrunkOne-0c12' The subsequent commands in the dialplan do not get processed. Can anyone suggest either why this would happen and how to get the dialplan to continue processing under all circumstances, or at least how to investigate futher what is causing this to happen? I'm sure that leaving a message and hanging up the call should be valid because that's what the default greeting message tells people they can do. Thanks, Antony. -- Why are they called "The Rocky Mountains"? What are other mountains made of? Please reply to the list; please *don't* CC me. -- _ -- 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] voicemail message not deleted
> Am 26.07.2021 um 07:28 schrieb Fourhundred Thecat <400the...@gmx.ch>: > > Hello, > > I have this in my voicemail.conf: > > attach=yes > > delete=yes > > I do get an email when new voicemail is received, and I do get the > voicemail message as attachment. > > However, the original message is not deleted from the sevber. > > How do I delete the message, after it has been sent per email as > attachment? I don't want to store messages on the server indefinitely. > > thanks, > > -- I think you need to set "delete=yes" as option per mailbox account. 100 => 1234,Test,,,delete=yes The global setting is only an example. Michael http://www.mksolutions.info -- _ -- 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
[asterisk-users] voicemail message not deleted
Hello, I have this in my voicemail.conf: attach=yes delete=yes I do get an email when new voicemail is received, and I do get the voicemail message as attachment. However, the original message is not deleted from the sevber. How do I delete the message, after it has been sent per email as attachment? I don't want to store messages on the server indefinitely. thanks, -- _ -- 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
[asterisk-users] Voicemail: don't play vm-intro if custom intro is recorded.
Hi Gang We migrated our voicemail system from asterisk 13 to 16 a couple of months ago. Right after the migration, we got the complaint that vm-intro is being played when the customer had recorded a own announcement. So I assumed we had replaced that file by a zero lenght one on the previous installation and did the same to suppress that surplus intro. Now I got the opposite complaint: If the customer did not record an own announcement, there is only the start of the into being played. The part "Please record your message after the tone" which resides in vm-intro is missing. I did try toggling the 's' option, but none fixes the behaviour. Any hint how I get back the previous behavior being: If customer recorded an own intro, only play the tone after the customer intro. If customer did not record an own intro, play the full intro. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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] Voicemail help when listening to messages
Hi, I have replicated this on a fresh 15.6.0 install, chan_sip, voicemail plain, vanilla config (whatever ships with the tucny.com RPMs) other than creating a SIP extension and voicemail user. Should I file a bug? > On 4/09/2018, at 3:46 PM, Nathan Ward wrote: > > Hi, > > I have a weird issue, unsure if it’s due to a bug, or configuration on my > end. We’re on 14.7.7. I’ve looked at the app_voicemail.c code, and see no > changes in this area of the code until the current version so don’t think age > of the code is an issue here (but happy to be proven wrong!). > > When hitting * for help when listening to messages (dial in, press 1, then > *), we get the following options: > > press 1 for new messages > press 2 to change folders > press 3 for advanced options > press 0 for mailbox options > press 1 to listen to new messages * > press 2 to access messages saved in other folders * > press 3 to record a message for another mailbox * > press 0 for greetings and password management * > press 5 to repeat the current message > press 6 to play the next message > press 7 to delete this message > press 8 to forward the message to another user > press 9 to save this message > press * for help or # to exit > > I’ve marked 4 items with *s, these are items which seem out of place, as the > other options largely work. > > I’ve been poking around, and it seems to be that this is happening because > `skipadvanced` is set to 1, though, it this is set to 0 it would (I think) be: > press 1 for new messages > press 2 to change folders > press 3 for advanced options > press 0 for mailbox options > press 3 for advanced options * > press 5 to repeat the current message > press 6 to play the next message > press 7 to delete this message > press 8 to forward the message to another user > press 9 to save this message > press * for help or # to exit > > Note the item with a * where advanced options is offered again (once from > vm-opts then later from vm-advopts). > > I am not clear on how this is supposed to work, as both with and without > skipadvanced set seems weird. Can anyone help? > > -- > Nathan Ward > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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
[asterisk-users] Voicemail help when listening to messages
Hi, I have a weird issue, unsure if it’s due to a bug, or configuration on my end. We’re on 14.7.7. I’ve looked at the app_voicemail.c code, and see no changes in this area of the code until the current version so don’t think age of the code is an issue here (but happy to be proven wrong!). When hitting * for help when listening to messages (dial in, press 1, then *), we get the following options: press 1 for new messages press 2 to change folders press 3 for advanced options press 0 for mailbox options press 1 to listen to new messages * press 2 to access messages saved in other folders * press 3 to record a message for another mailbox * press 0 for greetings and password management * press 5 to repeat the current message press 6 to play the next message press 7 to delete this message press 8 to forward the message to another user press 9 to save this message press * for help or # to exit I’ve marked 4 items with *s, these are items which seem out of place, as the other options largely work. I’ve been poking around, and it seems to be that this is happening because `skipadvanced` is set to 1, though, it this is set to 0 it would (I think) be: press 1 for new messages press 2 to change folders press 3 for advanced options press 0 for mailbox options press 3 for advanced options * press 5 to repeat the current message press 6 to play the next message press 7 to delete this message press 8 to forward the message to another user press 9 to save this message press * for help or # to exit Note the item with a * where advanced options is offered again (once from vm-opts then later from vm-advopts). I am not clear on how this is supposed to work, as both with and without skipadvanced set seems weird. Can anyone help? -- Nathan Ward -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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
[asterisk-users] Voicemail Directory
I am currently using Asterisk 13.21.1 under Ubuntu (Compiled from source). The Dial-by-name directory option that I'm currently using: Directory(sip,sip,eb) That allows for first and last name matching. I've recently enabled forwarding voicemail with the directory by enabling usedirectory=yes in voicemail.conf, but it only allows matching against last name. Is there a way to pass the 'b' parameter to the directory application so I can keep the options consistent? Thanks, Doug -- _ -- 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
[asterisk-users] Voicemail to emails
Hello all, I have setup the voicemail system with IMAP storage in asterisk 13.Using postfix MTA able to send the emails from the command line to email address provided but when a voicemail is arrived that is not being sent to emails address provided under voicemail.conf. Please anyone can help in this? -- _ -- 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] Voicemail: search for name in a phonebook
John Kinistonschrieb: > Yes, You could do easily this either with the internal asterisk database or > with something like func_odbc as a source for the data. > > In the context you receive your incoming calls you do a lookup against one > of the above data sources using the CALLERID(NUM) and change CALLERID(NAME) > to be the name you set. Thanks a lot! I found this page: http://deepliquid.com/blog/archives/59 and I successfully got it working! Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voicemail: search for name in a phonebook
Yes, You could do easily this either with the internal asterisk database or with something like func_odbc as a source for the data. In the context you receive your incoming calls you do a lookup against one of the above data sources using the CALLERID(NUM) and change CALLERID(NAME) to be the name you set. On Wed, Sep 20, 2017 at 1:04 PM, Luca Bertoncellowrote: > Hi list! > > I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly. > I configured a voicemail and I receive an E-Mail with some information > about > the call. > Again, wonderful! > > Now my wish: I'd like to have Asterisk to search the caller in a list file > and send me the name corresponding to the number in the E-Mail of > voicemail. > Is it possible? > > I currently use ${VM_CALLERID} in emailbody and it gives, of course, the > phone number... > > Thanks a lot! > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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
[asterisk-users] Voicemail: search for name in a phonebook
Hi list! I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly. I configured a voicemail and I receive an E-Mail with some information about the call. Again, wonderful! Now my wish: I'd like to have Asterisk to search the caller in a list file and send me the name corresponding to the number in the E-Mail of voicemail. Is it possible? I currently use ${VM_CALLERID} in emailbody and it gives, of course, the phone number... Thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voicemail asking for login
> "DC" == D'Arcy Cainwrites: DC> I did debug 10 and saved the console output into files which I DC> compared side by side. No material difference. In that case I'd add more debug statements to apps/app_voicemail.c (in vm_exec()), including a log at the start of what is in *data and args. Looking at it, it only plays vm-whichbox when ast_strlen_zero(data), which implies that the args to Voicemail are not making it through. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Voicemail asking for login
On 2017-04-20 04:07 PM, James Cloos wrote: I enable full log and run 'core set debug 9' before doing a pair of tests. (The full log is easier to grep than the console output.) Then compare a working vs stocktrans2 side by side. I did debug 10 and saved the console output into files which I compared side by side. No material difference. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
I enable full log and run 'core set debug 9' before doing a pair of tests. (The full log is easier to grep than the console output.) Then compare a working vs stocktrans2 side by side. -JimC -- James CloosOpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Voicemail asking for login
On 2017-04-20 12:52 PM, J Montoya wrote: On Thursday 20 Apr 2017, D'Arcy Cain wrote: On 2017-04-20 12:23 PM, D'Arcy Cain wrote: Here is the full dialplan for stocktrans2. I reduced this to the following and I still have the error. exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for stocktrans2) same => n,VoiceMail(stocktrans2@VoiceMail,u) same => n,Hangup() O.K., so -- assuming that extension "darcy" behaves properly if you simplify it similarly -- nothing before there can be causing the problem. Actually, it also failed too when I reduced it. Turns out that that's why I needed to set "_ACCOUNT". Here is the actual reduced version for stocktrans2. It still fails but substituting "darcy" for "stocktrans2 works OK. exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for stocktrans2) same => n,Set(_ACCOUNT=stocktrans2) same => n,VoiceMail(stocktrans2@VoiceMail,u) same => n,Hangup() What is in your [VoiceMail] context? Are "stocktrans2" and "darcy" separate extensions, or is there a catch-all? What is in the "a" extension (which gets called when the * key is pressed) ? Nothing in [VoiceMail] context except the mailboxes but this precedes the context: [general] attach=yes maxsilence=10 maxlogins=3 serveremail=n...@vex.net format=wav49 fromstring=Vybe Networks Voice Mail nextaftercmd=yes forcename=yes pollmailboxes=yes pollfreq=5 emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t-- Vybe Networks\n They are separate extensions. I do not do catch-alls if I can help it. Since I generate the configs it is no big deal to expand everything in each extension. ; voice mail exten => a,1,Verbose(${ACCOUNT} entering mailbox) same => n,Set(CDR(userfield)=${ACCOUNT}) same => n,VoicemailMain(${ACCOUNT}@VoiceMail) same => n,Hangup The VoiceMail extension, as are all the extensions, is in the [LocalSets] context. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
On Thursday 20 Apr 2017, D'Arcy Cain wrote: > On 2017-04-20 12:23 PM, D'Arcy Cain wrote: > > Here is the full dialplan for stocktrans2. > > I reduced this to the following and I still have the error. > > exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) > same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) > same => n,Verbose(0,${CALLERID(all)} going into voice mail for > stocktrans2) > same => n,VoiceMail(stocktrans2@VoiceMail,u) > same => n,Hangup() O.K., so -- assuming that extension "darcy" behaves properly if you simplify it similarly -- nothing before there can be causing the problem. What is in your [VoiceMail] context? Are "stocktrans2" and "darcy" separate extensions, or is there a catch-all? What is in the "a" extension (which gets called when the * key is pressed) ? -- JM Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail asking for login
On 2017-04-20 12:23 PM, D'Arcy Cain wrote: Here is the full dialplan for stocktrans2. I reduced this to the following and I still have the error. exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for stocktrans2) same => n,VoiceMail(stocktrans2@VoiceMail,u) same => n,Hangup() -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote: This is just screaming "configuration mismatch" -- or, possibly, "latent bug whereby things parsed in separate places should be treated the same, but are actually getting treated differently". I really don't want to be the "my system isn't working so there must be a bug in Asterisk" guy but I am certainly starting to suspect it. I think we are going to need to see your dialplan logic, and maybe your voicemail.conf, in order to work out what is different between this one user and all the others. You might even need to use `hd` to examine the files, just in case there is a stray non-printing character spoiling things. Non-printing characters show up in vi so I would have seen that by now. Every extension is build by a script that takes information from the database and does substitutions which is what makes this so baffling. Every extension is built exactly the same way. Here are two voicemail entries, the failing one and mine that works. The only sanitation I did was for the password. stocktrans2 => ,Angelica Douglas,stocktra...@vex.net darcy => ,Vybe Networks - D'Arcy,da...@vex.net Here is the full dialplan for stocktrans2. exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) same => n,Goto(DialCell) same => n,GotoIf($["x" = "x"]?DialAlt) same => n(DialAll),Verbose(0,${CALLERID(all)} Calling ${EXTEN} and ALL) same => n,Dial(SIP/stocktrans2/thinktel//907084,30) same => n,Goto(VoiceMail) same => n(DialAlt),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and SoftPhone) same => n,Dial(SIP/stocktrans2/907084,30) same => n,Goto(VoiceMail) same => n(DialCell),GotoIf($["x" = "x"]?DialDesk) same => n,GotoIf($["${CALLERID(ani)}" = ""]?DialDesk) same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell "") same => n,Dial(SIP/stocktrans2/thinktel/,30) same => n,Goto(VoiceMail) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/stocktrans2,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for stocktrans2) same => n,Set(_ACCOUNT=stocktrans2) same => n,VoiceMail(stocktrans2@VoiceMail,u) same => n,Hangup() Here is mine. exten => darcy,1,Verbose(0,Entering extension darcy) same => n,GotoIf($["${DEVICE_STATE(SIP/901001)}" = "UNAVAILABLE"]?DialCell) same => n,GotoIf($["x4168035991" = "x"]?DialAlt) same => n(DialAll),Verbose(0,${CALLERID(all)} Calling ${EXTEN} and ALL) same => n,Dial(SIP/darcy/thinktel/4168035991/901001,30) same => n,Goto(VoiceMail) same => n(DialAlt),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and SoftPhone) same => n,Dial(SIP/darcy/901001,30) same => n,Goto(VoiceMail) same => n(DialCell),GotoIf($["x4168035991" = "x"]?DialDesk) same => n,GotoIf($["${CALLERID(ani)}" = "4168035991"]?DialDesk) same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell "4168035991") same => n,Dial(SIP/darcy/thinktel/4168035991,30) same => n,Goto(VoiceMail) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/darcy,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for darcy) same => n,Set(_ACCOUNT=darcy) same => n,VoiceMail(darcy@VoiceMail,u) same => n,Hangup() There are some minor differences based on whether they have set up an alternate phone but either way it gets to the line that set the CDR userfield. They both ultimately hit one or the other of these lines. same => n,VoiceMail(stocktrans2@VoiceMail,u) same => n,VoiceMail(darcy@VoiceMail,u) The only "Set" command that might change the environment is the setting of "_ACCOUNT" which they both do. Not sure why I even do that. Perhaps I was planning some other feature that I never finished. Or does VoiceMail() use it? Someone (in private email for some reason so I won't give his name) suggested DumpChan() but I can't seem to make that work, even if I load app_dumpchan. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
On Wednesday 19 Apr 2017, D'Arcy Cain wrote: > Yes and [using something like "1571"] works just fine for us. The problem > is that we are trying > to deal with the situation where someone calls themselves from another > phone (internal or external) to pick up their messages. In every other > case it asks for their password (which is always numeric) and goes into > the VM. This one extension asks for a mailbox. This is just screaming "configuration mismatch" -- or, possibly, "latent bug whereby things parsed in separate places should be treated the same, but are actually getting treated differently". I think we are going to need to see your dialplan logic, and maybe your voicemail.conf, in order to work out what is different between this one user and all the others. You might even need to use `hd` to examine the files, just in case there is a stray non-printing character spoiling things. -- JM or AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Voicemail asking for login
On 2017-04-19 11:57 AM, J Montoya or A J Stiles wrote: I fished this out of an old extensions.conf from a defunct project. It might be relevant to your use case: exten => 1571,1,NoOp(Call to 1571: voicemail retrieval) exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)}) exten => 1571,n,NoOp(CLID is ${clid}) exten => 1571,n,VoiceMailMain(${clid},s) I do something similar using *98. The upshot of this was, if you dialled 1571 from your own phone, then you got put straight through to your own voicemail, without logging in. Yes and that works just fine for us. The problem is that we are trying to deal with the situation where someone calls themselves from another phone (internal or external) to pick up their messages. In every other case it asks for their password (which is always numeric) and goes into the VM. This one extension asks for a mailbox. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
On Wednesday 19 Apr 2017, D'Arcy Cain wrote: > On 2017-04-19 02:39 AM, Pete Mundy wrote: > > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail > > > > show users' I can't see why the vm_authenticate function is > > failing to read the username :( > > I can answer that one. It's because we can't enter 'stocktrans2' from a > telephone so we just hang up. The question is, why does it ask for the > mailbox in the first place> I fished this out of an old extensions.conf from a defunct project. It might be relevant to your use case: exten => 1571,1,NoOp(Call to 1571: voicemail retrieval) exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)}) exten => 1571,n,NoOp(CLID is ${clid}) exten => 1571,n,VoiceMailMain(${clid},s) The AGI script `lookup_caller_id.agi` sets the variable ${clid} to the caller's extension number, after which their mailbox is named (although there is no reason not to set another variable, such as ${mbox} to hold the mailbox if you want). In the call to voicemailmail() we specify this mailbox, and also use the `s` option to skip password checking (it was safe in this situation to assume that nobody had physical access to a phone who definitely should not have had access to its user's voicemail messages). The upshot of this was, if you dialled 1571 from your own phone, then you got put straight through to your own voicemail, without logging in. -- JM or AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Voicemail asking for login
On 2017-04-19 02:39 AM, Pete Mundy wrote: Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail > show users' I can't see why the vm_authenticate function is > failing to read the username :( I can answer that one. It's because we can't enter 'stocktrans2' from a telephone so we just hang up. The question is, why does it ask for the mailbox in the first place> If I were any good at coding in C, I'd probably look inside > app_voicemail.c around line number 10671 and see if I could > determine how it reads the username and maybe throw some hacky ? debug output in there to try and determine at which point of > that process it's failing. But I'm no good at coding in that > language, so will have to defer to others to help. I guess that's my next stop. Luckily I have kept my C skills somewhat active as the chief maintainer for PyGreSQL. :-) -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
> On 19/04/2017, at 4:25 pm, D'Arcy Cainwrote: > >> Does this mailbox exist? > > Yes. Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail show users' I can't see why the vm_authenticate function is failing to read the username :( If I were any good at coding in C, I'd probably look inside app_voicemail.c around line number 10671 and see if I could determine how it reads the username and maybe throw some hacky debug output in there to try and determine at which point of that process it's failing. But I'm no good at coding in that language, so will have to defer to others to help. Good work on sending through the console clipping and relevant info. Sorry I couldn't resolve it for you. Anyone else got any other ideas? Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Voicemail asking for login
On 2017-04-18 08:17 PM, Pete Mundy wrote: On 19/04/2017, at 7:58 am, D'Arcy Cain> wrote: Everything looks the same as another one that works except for two things. The one that works doesn't have the "Probation passed" lines. I am not sure if that is even part of this call. The other is the line with "Playing 'vm-login.gsm'" in it. at that point the working one has this: Presumably also the line containing 'vm_authenticate: Couldn't read username' also doesn't appear in the output on a working mailbox either? Exactly. Since it is not all digits it can't be entered. I think that's the place to concentrate your efforts. It shows shortly after the attempt by VoiceMailMain to enter mailbox 'stocktrans2' in context 'VoiceMail'. Does this mailbox exist? Yes. Can you show the equivalent line from a working mailbox (so we can see if it also uses the context 'VoiceMail', or maybe something else instead, like 'default'?). "" <6477190146> going into voice mail for alex<<< -- Executing [alex@LocalSets:19] Set("SIP/thinktel-0181", "_ACCOUNT=alex") in new stack<<< -- Executing [alex@LocalSets:20] VoiceMail("SIP/thinktel-0181", "alex@VoiceMail,u") in new stack<<< -- Playing '/var/spool/asterisk/voicemail/VoiceMail/alex/unavail.gsm' (language 'en')<<< [Apr 18 11:56:47] DTMF[-1][C-0004c485]: channel.c:4215 __ast_read: DTMF begin '*' received on SIP/thinktel-0181<<< [Apr 18 11:56:47] DTMF[-1][C-0004c485]: channel.c:4219 __ast_read: DTMF begin ignored '*' on SIP/thinktel-0181<<< [Apr 18 11:56:48] DTMF[-1][C-0004c485]: channel.c:4129 __ast_read: DTMF end '*' received on SIP/thinktel-0181, duration 280 ms<<< [Apr 18 11:56:48] DTMF[-1][C-0004c485]: channel.c:4199 __ast_read: DTMF end passthrough '*' on SIP/thinktel-0181<<< -- Executing [a@LocalSets:1] Verbose("SIP/thinktel-0181", "alex entering mailbox") in new stack<<< alex entering mailbox<<< -- Executing [a@LocalSets:2] Set("SIP/thinktel-0181", "CDR(userfield)=alex") in new stack<<< -- Executing [a@LocalSets:3] VoiceMailMain("SIP/thinktel-0181", "alex@VoiceMail") in new stack<<< -- Playing 'vm-password.gsm' (language 'en')<<< [Apr 18 11:56:53] WARNING[-1][C-0004c485]: app_voicemail.c:10671 vm_authenticate: Unable to read password<<< I hung up before entering the password but it does work when the user does it. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
On 2017-04-18 08:31 PM, Victor Villarreal wrote: Maybe excecuting the following command at Asterisk console, will help you: asterisk> voicemail show users And you will get a list of all mailbox configured in your system. Search for the user with problems. VoiceMail stocktrans2 Angelica Douglas 12 Definitely there. In fact, I generate all the configs from a database with a script so I would be very surprised if one user was different from another. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail asking for login
Hi Darcy, What Pete think is correct. Maybe excecuting the following command at Asterisk console, will help you: asterisk> voicemail show users And you will get a list of all mailbox configured in your system. Search for the user with problems. Finally, in the Asterisk wiki you can find more info: https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes Cheers El 18 abr. 2017 21:18, "Pete Mundy"escribió: On 19/04/2017, at 7:58 am, D'Arcy Cain wrote: Everything looks the same as another one that works except for two things. The one that works doesn't have the "Probation passed" lines. I am not sure if that is even part of this call. The other is the line with "Playing 'vm-login.gsm'" in it. at that point the working one has this: Presumably also the line containing 'vm_authenticate: Couldn't read username' also doesn't appear in the output on a working mailbox either? I think that's the place to concentrate your efforts. It shows shortly after the attempt by VoiceMailMain to enter mailbox 'stocktrans2' in context 'VoiceMail'. Does this mailbox exist? Can you show the equivalent line from a working mailbox (so we can see if it also uses the context 'VoiceMail', or maybe something else instead, like 'default'?). Pete -- _ -- 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 -- _ -- 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] Voicemail asking for login
> On 19/04/2017, at 7:58 am, D'Arcy Cainwrote: > > > Everything looks the same as another one that works except for two things. > The one that works doesn't have the "Probation passed" lines. I am not sure > if that is even part of this call. The other is the line with "Playing > 'vm-login.gsm'" in it. at that point the working one has this: > Presumably also the line containing 'vm_authenticate: Couldn't read username' also doesn't appear in the output on a working mailbox either? I think that's the place to concentrate your efforts. It shows shortly after the attempt by VoiceMailMain to enter mailbox 'stocktrans2' in context 'VoiceMail'. Does this mailbox exist? Can you show the equivalent line from a working mailbox (so we can see if it also uses the context 'VoiceMail', or maybe something else instead, like 'default'?). Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Voicemail asking for login
On 2017-04-18 02:42 AM, Pete Mundy wrote: Try this: asterisk -r core set verbose 10 [get user to trigger fault] [examine console output, and post to list if still unclear] If you don't solve it yourself, then we'll be able to help further once we've seen the output. I can't see much more than at my previous debug level but here it is anyway. Due to line wrapping I added "<<<" to the end of each line in case it is not clear where the actual line endings are. "Alex Chernyshev" <4164251212> going into voice mail for stocktrans2<<< -- Executing [stocktrans2@LocalSets:19] Set("SIP/alex-0175", "_ACCOUNT=stocktrans2") in new stack<<< -- Executing [stocktrans2@LocalSets:20] VoiceMail("SIP/alex-0175", "stocktrans2@VoiceMail,u") in new stack<<< > 0x7f7fea5dc000 -- Probation passed - setting RTP source address to 72.143.94.110:28503<<< -- Playing '/var/spool/asterisk/voicemail/VoiceMail/stocktrans2/unavail.gsm' (language 'en')<<< > 0x7f7fea5dc000 -- Probation passed - setting RTP source address to 72.143.94.110:28503<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4215 __ast_read: DTMF begin '*' received on SIP/alex-0175<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4219 __ast_read: DTMF begin ignored '*' on SIP/alex-0175<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4129 __ast_read: DTMF end '*' received on SIP/alex-0175, duration 160 ms<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4199 __ast_read: DTMF end passthrough '*' on SIP/alex-0175<<< -- Executing [a@LocalSets:1] Verbose("SIP/alex-0175", "stocktrans2 entering mailbox") in new stack<<< stocktrans2 entering mailbox<<< -- Executing [a@LocalSets:2] Set("SIP/alex-0175", "CDR(userfield)=stocktrans2") in new stack<<< -- Executing [a@LocalSets:3] VoiceMailMain("SIP/alex-0175", "stocktrans2@VoiceMail") in new stack<<< -- Playing 'vm-login.gsm' (language 'en')<<< [Apr 18 11:45:49] WARNING[-1][C-0004c47b]: app_voicemail.c:10627 vm_authenticate: Couldn't read username<<< Everything looks the same as another one that works except for two things. The one that works doesn't have the "Probation passed" lines. I am not sure if that is even part of this call. The other is the line with "Playing 'vm-login.gsm'" in it. at that point the working one has this: -- Playing 'vm-password.gsm' (language 'en') Not sure if that's useful information since it just describes the original issue - that it asks for a login instead of a password. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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] Voicemail asking for login
Does he have the same voicemail context?From: p...@fiberphone.co.nzSent: April 18, 2017 9:43 AMTo: asterisk-users@lists.digium.comReply-to: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Voicemail asking for login Hi D'ArcyOn 18/04/2017, at 5:17 am, D'Arcy Cain <da...@vybenetworks.com> wrote:One user (that we know of so far) has a different experience. In that case they are asked for a mailbox number first. I have tried searching for this issue but nothing seems to apply. Most discussions are about "*97" vs. "*98". Can anyone suggest another field of enquiry?Try this: asterisk -r core set verbose 10 [get user to trigger fault] [examine console output, and post to list if still unclear]If you don't solve it yourself, then we'll be able to help further once we've seen the output.HTH,Pete-- _ -- 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] Voicemail asking for login
Hi D'Arcy > On 18/04/2017, at 5:17 am, D'Arcy Cainwrote: > > > One user (that we know of so far) has a different experience. In that case > they are asked for a mailbox number first. > > I have tried searching for this issue but nothing seems to apply. Most > discussions are about "*97" vs. "*98". Can anyone suggest another field of > enquiry? Try this: asterisk -r core set verbose 10 [get user to trigger fault] [examine console output, and post to list if still unclear] If you don't solve it yourself, then we'll be able to help further once we've seen the output. HTH, Pete smime.p7s Description: S/MIME cryptographic signature -- _ -- 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
[asterisk-users] Voicemail asking for login
We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%) same => n,Set(_ACCOUNT=%ACCOUNT%) same => n,VoiceMail(%ACCOUNT%@VoiceMail,u) same => n,Hangup() And for voicemail.conf: %ACCOUNT% => %VM_PASSWORD%,%NAME%,%log...@vex.net Here is the sip.conf template: [%ACCOUNT%](client-phone) secret=%PASSWORD% callerid=%NAME% <%CLID%> mailbox=%ACCOUNT%@VoiceMail context=%CONTEXT% Every user gets set up using these templates so I know that everyone is identical other than the '%' variables above. I have looked and I don't see any significant differences. The ACCOUNTs are strings with most having digits appended. Obviously NAME, PASSWORD and LOGIN are different but not in kind. My issue is with users picking up their VM from an external phone. They call themselves and press '*' during the playback message. Normally they are asked for their password and then get dropped into the proper menu. One user (that we know of so far) has a different experience. In that case they are asked for a mailbox number first. I can't seem to find any significant difference in their configuration to account for that. Every other user that we have tested works as expected. Some of them have extension that are all letters, some have trailing digits. Some have associated cell phones and some don't. I have tried searching for this issue but nothing seems to apply. Most discussions are about "*97" vs. "*98". Can anyone suggest another field of enquiry? TIA. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Voicemail notification by email is missing CallerID info
I’ll go through it and see what I missed. I can't thank you enough! John From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H Sent: Saturday, February 18, 2017 11:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail notification by email is missing CallerID info This is what comes with voicemail.conf.sample - works for me! ; Change the from, body and/or subject, variables: ; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, ; VM_CIDNAME, VM_DATE ; Additionally, on forwarded messages, you have the variables: ; ORIG_VM_CALLERID, ORIG_VM_CIDNUM, ORIG_VM_CIDNAME, ORIG_VM_DATE ; You can select between two variables by using dialplan functions, e.g. ; ${IF(${ISNULL(${ORIG_VM_DATE})}?${VM_DATE}:${ORIG_VM_DATE})} ; ; Note: The emailbody config row can only be up to 512 characters due to a ; limitation in the Asterisk configuration subsystem. ;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX} ; The following definition is very close to the default, but the default shows ; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown ; caller", if they are both null. ;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n ; ; Note: ${IF()} strips spacing at the beginning and end of its true and false ; values, so a newline cannot be placed at either location. The word 'so' is ; therefore duplicated, in order for the newline to be interpreted correctly. ;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just ${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n On 18 February 2017 at 16:35, Tech Support <aster...@voipbusiness.us> wrote: > All; > > I am running Asterisk 11.6-cert16 and I have voicemail setup so > voicemail messages are sent as email attachments. That works fine. However, > the body of the email contains the CallerID(name), but is missing the > CallerID(num). For example, the email body looks like this: > > > > Just wanted to let you know you were just left a 0:21 long message > (number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17, > 2017 at 04:48:38 PM so you might want to check it when you get a chance. > Thanks! > > > > Checking the CDR’s shows that both the name and number were recorded by > Asterisk. Am I missing something obvious? Is it a simple config option in > voicemail.conf? Any insight at all would be greatly appreciated. > > Thanks; > > John V. > > > > > -- > _ > -- 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 -- _ -- 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] Voicemail notification by email is missing CallerID info
This is what comes with voicemail.conf.sample - works for me! ; Change the from, body and/or subject, variables: ; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, ; VM_CIDNAME, VM_DATE ; Additionally, on forwarded messages, you have the variables: ; ORIG_VM_CALLERID, ORIG_VM_CIDNUM, ORIG_VM_CIDNAME, ORIG_VM_DATE ; You can select between two variables by using dialplan functions, e.g. ; ${IF(${ISNULL(${ORIG_VM_DATE})}?${VM_DATE}:${ORIG_VM_DATE})} ; ; Note: The emailbody config row can only be up to 512 characters due to a ; limitation in the Asterisk configuration subsystem. ;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX} ; The following definition is very close to the default, but the default shows ; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown ; caller", if they are both null. ;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n ; ; Note: ${IF()} strips spacing at the beginning and end of its true and false ; values, so a newline cannot be placed at either location. The word 'so' is ; therefore duplicated, in order for the newline to be interpreted correctly. ;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just ${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n On 18 February 2017 at 16:35, Tech Supportwrote: > All; > > I am running Asterisk 11.6-cert16 and I have voicemail setup so > voicemail messages are sent as email attachments. That works fine. However, > the body of the email contains the CallerID(name), but is missing the > CallerID(num). For example, the email body looks like this: > > > > Just wanted to let you know you were just left a 0:21 long message > (number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17, > 2017 at 04:48:38 PM so you might want to check it when you get a chance. > Thanks! > > > > Checking the CDR’s shows that both the name and number were recorded by > Asterisk. Am I missing something obvious? Is it a simple config option in > voicemail.conf? Any insight at all would be greatly appreciated. > > Thanks; > > John V. > > > > > -- > _ > -- 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 -- _ -- 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
[asterisk-users] Voicemail notification by email is missing CallerID info
All; I am running Asterisk 11.6-cert16 and I have voicemail setup so voicemail messages are sent as email attachments. That works fine. However, the body of the email contains the CallerID(name), but is missing the CallerID(num). For example, the email body looks like this: Just wanted to let you know you were just left a 0:21 long message (number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17, 2017 at 04:48:38 PM so you might want to check it when you get a chance. Thanks! Checking the CDR's shows that both the name and number were recorded by Asterisk. Am I missing something obvious? Is it a simple config option in voicemail.conf? Any insight at all would be greatly appreciated. Thanks; John V. -- _ -- 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] voicemail greeting
> hi.i managed to record my voicemail greeting. the only problem is that after > my greeting the caller hear '...please leave your message after the tone. > when done press the pound key or hangup.' is there a way to get rid of that? > Ideally i would like to have my own recording and then the beep sound. > Try option s : https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail Regards, -- Bertrand LUPART -- _ -- 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
[asterisk-users] voicemail greeting
hi.i managed to record my voicemail greeting. the only problem is that after my greeting the caller hear '...please leave your message after the tone. when done press the pound key or hangup.' is there a way to get rid of that? Ideally i would like to have my own recording and then the beep sound. -- _ -- 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] VoiceMail - Allow * for only some users
Hi John, Ah ha! Excellent. That works. Now for a further tweak, in my stdexten I set voicemail_option with with b or u, as appropriate and use ${voicemail_option) instead of option in the call to Voicemail below so the correct prompt is used. Thank you! On Thu, 2016-07-21 at 14:53 -0700, John Kiniston wrote: > I think you almost have it. > > In your vmfwd context have a wildcard match that sends the caller > back to the originating voicemail and then define specific extensions > that are allowed to forward. > > > [vmfwd] > exten => _,1,Voicemail(box@context,option) > same => n,Hangup > > ; Andrew Ruthven > exten => 7231,1,Set(CALLERID(number)=yyy) > same => n,Goto(pstn,xxx,1) > > On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthvenyst.net.nz> wrote: > > Hey, > > > > I have free calling to between DDIs and cellphones on our group > > plan. I > > figure it'd be nice to allow staff with those cellphones to be able > > to > > forward callers to their VoiceMail to their cellphones using the * > > feature. > > > > I have a standard extension macro that has VoiceMail support. > > So far I've done this by duplicating the standard extension macro, > > and > > adding this rule (where ARG1 is the extension): > > > > exten => a,1,Goto(vmfwd,${ARG1},1) > > > > Then in the vmfwd context I have rules like this (I need to set the > > CALLERID(number) so our SIP provider accepts the call): > > > > ; Andrew Ruthven > > exten => 7231,1,Set(CALLERID(number)=yyy) > > exten => 7231,n,Goto(pstn,xxx,1) > > > > Which is working nicely. But, I thought, can I simplify this and > > just > > have one macro? > > > > So I've tried doing the following to fold it into my standard > > extension > > macro: > > > > 1) Tried using a/_7231 but that didn't match (well, it was worth a > > try) > > 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my > > extension, > > but if I disable the 7231 rules in vmfwd, I get: > > > > [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646 > > __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to > > invalid > > extension but no invalid handler: > > context,exten,priority=vmfwd,7231,1 > > > > and the call hangs up, not a very nice user experience. > > > > The second option could work, as long as the user lands back into > > VoiceMail if there is no valid extension. I thought about using > > GoSub, > > but how do I get the caller back into VoiceMail? > > > > I've done a bunch of searching for this, but haven't found any > > general > > solutions. Is it possible to do what I'm trying to achieve, or is > > there > > a better approach? > > > > This is Asterisk 11.13. > > > > Cheers, > > Andrew > > > > -- > > > > Andrew Ruthven, Wellington, New Zealand > > MIITP, CITPNZ > > > > At work: andrew.ruth...@catalyst.net.nz > > At home: and...@etc.gen.nz > > Card : http://qr.catalyst.net.nz/907675e1 > > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud > > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 > > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org > > > > > > > > > > > > -- > > ___ > > __ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > > -- > > New to Asterisk? Join us for a live introductory webinar every > > Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- Andrew Ruthven, Wellington, New Zealand MIITP, CITPNZ At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Card : http://qr.catalyst.net.nz/907675e1 Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail - Allow * for only some users
I think you almost have it. In your vmfwd context have a wildcard match that sends the caller back to the originating voicemail and then define specific extensions that are allowed to forward. [vmfwd] exten => _,1,Voicemail(box@context,option) same => n,Hangup ; Andrew Ruthven exten => 7231,1,Set(CALLERID(number)=yyy) same => n,Goto(pstn,xxx,1) On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven < andrew.ruth...@catalyst.net.nz> wrote: > Hey, > > I have free calling to between DDIs and cellphones on our group plan. I > figure it'd be nice to allow staff with those cellphones to be able to > forward callers to their VoiceMail to their cellphones using the * > feature. > > I have a standard extension macro that has VoiceMail support. > So far I've done this by duplicating the standard extension macro, and > adding this rule (where ARG1 is the extension): > > exten => a,1,Goto(vmfwd,${ARG1},1) > > Then in the vmfwd context I have rules like this (I need to set the > CALLERID(number) so our SIP provider accepts the call): > > ; Andrew Ruthven > exten => 7231,1,Set(CALLERID(number)=yyy) > exten => 7231,n,Goto(pstn,xxx,1) > > Which is working nicely. But, I thought, can I simplify this and just > have one macro? > > So I've tried doing the following to fold it into my standard extension > macro: > > 1) Tried using a/_7231 but that didn't match (well, it was worth a try) > 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension, > but if I disable the 7231 rules in vmfwd, I get: > > [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646 > __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid > extension but no invalid handler: context,exten,priority=vmfwd,7231,1 > > and the call hangs up, not a very nice user experience. > > The second option could work, as long as the user lands back into > VoiceMail if there is no valid extension. I thought about using GoSub, > but how do I get the caller back into VoiceMail? > > I've done a bunch of searching for this, but haven't found any general > solutions. Is it possible to do what I'm trying to achieve, or is there > a better approach? > > This is Asterisk 11.13. > > Cheers, > Andrew > > -- > > Andrew Ruthven, Wellington, New Zealand > MIITP, CITPNZ > > At work: andrew.ruth...@catalyst.net.nz > At home: and...@etc.gen.nz > Card : http://qr.catalyst.net.nz/907675e1 > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] VoiceMail - Allow * for only some users
Hey, I have free calling to between DDIs and cellphones on our group plan. I figure it'd be nice to allow staff with those cellphones to be able to forward callers to their VoiceMail to their cellphones using the * feature. I have a standard extension macro that has VoiceMail support. So far I've done this by duplicating the standard extension macro, and adding this rule (where ARG1 is the extension): exten => a,1,Goto(vmfwd,${ARG1},1) Then in the vmfwd context I have rules like this (I need to set the CALLERID(number) so our SIP provider accepts the call): ; Andrew Ruthven exten => 7231,1,Set(CALLERID(number)=yyy) exten => 7231,n,Goto(pstn,xxx,1) Which is working nicely. But, I thought, can I simplify this and just have one macro? So I've tried doing the following to fold it into my standard extension macro: 1) Tried using a/_7231 but that didn't match (well, it was worth a try) 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension, but if I disable the 7231 rules in vmfwd, I get: [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646 __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid extension but no invalid handler: context,exten,priority=vmfwd,7231,1 and the call hangs up, not a very nice user experience. The second option could work, as long as the user lands back into VoiceMail if there is no valid extension. I thought about using GoSub, but how do I get the caller back into VoiceMail? I've done a bunch of searching for this, but haven't found any general solutions. Is it possible to do what I'm trying to achieve, or is there a better approach? This is Asterisk 11.13. Cheers, Andrew -- Andrew Ruthven, Wellington, New Zealand MIITP, CITPNZ At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Card : http://qr.catalyst.net.nz/907675e1 Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail and SMS
On Friday 15 Jul 2016, Joaquin Alzola wrote: > Hi Guys > > I am asking too many questions because we would like to use Asterisk first > as a proof of Concept and check from there were it goes. > > - Does the Voicemail have the option of SMS notification on new drop > messages (we have an SMSC so we will use that one). Asterisk Voicemail can certainly send an e-mail when a message is left. By cunning use of a procmail recipe, this can be used to send an SMS or do anything else. > - What is the best Linux OS to install Asterisk in? The one with which you are most familiar. > - What throughput does it stand 1 machine with about 8GB Ram and 4 CPUs? We > plan to add couple but just checking for a single one. I've seen boxes with 2 cores, 4 GB RAM, 8 outside lines, all calls recorded using MixMonitor and no swapping; 4 cores, 8 GB RAM, 20 outside lines on an ISDN30 and more via SIP trunks, MixMonitor recording and again no swapping. > - Does it hava a max capacity? Probably, but good luck trying to find it :) -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail and SMS
How many users are you thinking of supporting? For a large-scale setup you might want to take a look at Kamailio as a front-end - if you even think you're going to get a high user volume you may want to start out with a Kamailio front-end so that you don't have to start over from scratch when it outgrows an Asterisk only setup. I prefer Debian/Ubuntu over RedHat/CentOS - but if you're thinking of using this in a company environment, I'd recommend engaging with your IT people to find out what THEIR preference is. Planning capacity with the information you've provided is difficult - is the network card 10Base-T, Gigabit, 100-gigabit? Do you only have 56kbps dial-up service to the server, or a full 10Gbps internet connection at a carrier-neutral colocation datacenter on a fiber backbone? Are the CPU cores 15-year-old Pentium or a current Broadwell-E? Are the CPU cores real or on a massively over-provisioned VM host? Do you have to do a bunch of transcoding inbound and outbound? Is the machine doing anything other than voicemail? In general, the maximum capacity is the point just before when the quality begins to drop (about 90-95% total system load). Not very scientific I know, but the answer is extremely hardware/infrastructure/setup dependent. I haven't personally played with any of Asterisk's internal SMS functionality, but I have been meaning to. Since the earlier days I've relied on the email functions to handle interfacing notifications. Asterisk can send emails as a notification, so I configured the default email to a notification handler, which would do a speech recognition on the voicemail file, send an SMS using an SMSC (Nexmo in my case), and then send an email to the user with a text transcript of the voicemail as well as the audio file as an attachment. I'm sure there's a better way now, I coded this up a while ago. My way is probably not the "right way", but like many things with computers there is the way that works today, the way that works better tomorrow, and eventually the best practice way that emerges after a few years. Gotta keep maintaining your work. -Tim On Fri, Jul 15, 2016 at 8:29 AM, Joaquin Alzolawrote: > Hi Guys > > I am asking too many questions because we would like to use Asterisk first > as a proof of Concept and check from there were it goes. > > - Does the Voicemail have the option of SMS notification on new drop > messages (we have an SMSC so we will use that one). > - What is the best Linux OS to install Asterisk in? > - What throughput does it stand 1 machine with about 8GB Ram and 4 CPUs? > We plan to add couple but just checking for a single one. > - Does it have a max capacity? > > Thanks for your time. > > BR > > Joaquin > This email is confidential and may be subject to privilege. If you are not > the intended recipient, please do not copy or disclose its content but > contact the sender immediately upon receipt. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail and SMS
Hi Guys I am asking too many questions because we would like to use Asterisk first as a proof of Concept and check from there were it goes. - Does the Voicemail have the option of SMS notification on new drop messages (we have an SMSC so we will use that one). - What is the best Linux OS to install Asterisk in? - What throughput does it stand 1 machine with about 8GB Ram and 4 CPUs? We plan to add couple but just checking for a single one. - Does it hava a max capacity? Thanks for your time. BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] VoiceMail Audio playing
> No. The VoiceMail server takes care of all that itself; it delivers the > broadcast and records the messages. Thanks AJ. This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] VoiceMail Audio playing
On Friday 15 Jul 2016, Joaquin Alzola wrote: > Hi Madushan > > Maybe I was not clear …. After SIP negotiation and SDP set up on the > VoiceMail Server …. > > Is there a file to specify a MGw (the machine that deliver RTP packages to > end user)? No. The VoiceMail server takes care of all that itself; it delivers the broadcast and records the messages. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Audio playing
> Asterisk does not separate things like this. For media originating from it > the source will always be it. That is if you do a SIP call to Asterisk then > media will come from that same Asterisk. Joshua ok perfect so Asterisk already have the play module incorporated. That’s great to hear so no need to integrate it to a MediaGatwey or SBC. This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] VoiceMail Audio playing
Joaquin Alzola wrote: Hi Madushan Maybe I was not clear …. After SIP negotiation and SDP set up on the VoiceMail Server …. Is there a file to specify a MGw (the machine that deliver RTP packages to end user)? Asterisk does not separate things like this. For media originating from it the source will always be it. That is if you do a SIP call to Asterisk then media will come from that same Asterisk. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Audio playing
Hi Madushan Maybe I was not clear …. After SIP negotiation and SDP set up on the VoiceMail Server …. Is there a file to specify a MGw (the machine that deliver RTP packages to end user)? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Madushan Geethanga Sent: 15 July 2016 13:00 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] VoiceMail Audio playing Hi, VoiceMailMain is used to retrieve voice mails http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain Best Regards, Madushan On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzola <joaquin.alz...@lebara.com<mailto:joaquin.alz...@lebara.com>> wrote: Hi Guys Which module on Asterisk is the one in charge of playing the VoiceMail Server Audio to the end customer? I have work with MRFP but is it a module included in the SW? Need and external source? BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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 This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] VoiceMail Audio playing
Hi, VoiceMailMain is used to retrieve voice mails http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain Best Regards, Madushan On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzolawrote: > Hi Guys > > > > Which module on Asterisk is the one in charge of playing the VoiceMail > Server Audio to the end customer? > > I have work with MRFP but is it a module included in the SW? Need and > external source? > > > > BR > > > > Joaquin > This email is confidential and may be subject to privilege. If you are not > the intended recipient, please do not copy or disclose its content but > contact the sender immediately upon receipt. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail Audio playing
Hi Guys Which module on Asterisk is the one in charge of playing the VoiceMail Server Audio to the end customer? I have work with MRFP but is it a module included in the SW? Need and external source? BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] Voicemail Mailboxes + Cassandra
Hi List I have two questions: 1- Mailbox on the Asterisk Voicemail Server are created automatically? 2- Is there any support on the code to put the voice records on a Cassandra NoSQL database? BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] voicemail: duration while leaving a message
Thank you joshua.Le 9 mai 2016 à 16:00, Joshua Colpa écrit :Mamadou NGOM wrote:Hello list,Kia ora,I am asking when a caller want to leave a message to a mailbox with theapplication voicemailHow i can limit the duration for exemple 30 seconds.exten => _X,n,VoiceMail(${Caller_number},s)Is there a option which allows me to do it, somebody to help me.This can be configured in voicemail.conf using the "maxsecs" configuration option. I don't believe this is exposed using the Voicemail application options, just using the config file.Cheers,-- Joshua ColpDigium, Inc. | Senior Software Developer445 Jan Davis Drive NW - Huntsville, AL 35806 - USCheck us out at: www.digium.com & www.asterisk.org-- _-- Bandwidth and Colocation Provided by http://www.api-digital.com --New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersMamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06 72 45 23 03Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 -- _ -- 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] voicemail: duration while leaving a message
Mamadou NGOM wrote: Hello list, Kia ora, I am asking when a caller want to leave a message to a mailbox with the application voicemail How i can limit the duration for exemple 30 seconds. exten => _X,n,VoiceMail(${Caller_number},s) Is there a option which allows me to do it, somebody to help me. This can be configured in voicemail.conf using the "maxsecs" configuration option. I don't believe this is exposed using the Voicemail application options, just using the config file. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail: duration while leaving a message
Hello list,I am asking when a caller want to leave a message to a mailbox with the application voicemailHow i can limit the duration for exemple 30 seconds.exten => _X,n,VoiceMail(${Caller_number},s)Is there a option which allows me to do it, somebody to help me.Best regards !!!Mamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06 72 45 23 03Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 -- _ -- 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] Voicemail using object storage?
I'd say using s3fs (or similar) is an approach, but if VoiceMail had support baked into it for S3, then the integration would be better. I'll look into using one the FUSE based approaches as a stop-gap measure. ;) On Tue, 2016-02-16 at 13:12 +0100, Olivier wrote: > Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be > mounted on Linux and thus allow any application like Asterisk make > use of it ? > > [1] https://github.com/s3fs-fuse/s3fs-fuse > > 2016-02-16 1:05 GMT+01:00 Andrew Ruthven.nz>: > > Hey, > > > > I've found a bit of chatter about people using hacks to copy > > voicemail > > messages into object storage (like S3) after they've been recorded. > > But > > I was wondering if any work has been done on the VoiceMail app to > > actually store and retrieve messages to/from an object store? > > > > Cheers, > > Andrew > > -- > > Andrew Ruthven, Wellington, New Zealand > > MIITP, ITCP > > > > At work: andrew.ruth...@catalyst.net.nz > > At home: and...@etc.gen.nz > > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud > > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 > > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org > > > > > > > > -- > > ___ > > __ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > > -- > > New to Asterisk? Join us for a live introductory webinar every > > Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew Ruthven, Wellington, New Zealand MIITP, ITCP At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail using object storage?
Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be mounted on Linux and thus allow any application like Asterisk make use of it ? [1] https://github.com/s3fs-fuse/s3fs-fuse 2016-02-16 1:05 GMT+01:00 Andrew Ruthven: > Hey, > > I've found a bit of chatter about people using hacks to copy voicemail > messages into object storage (like S3) after they've been recorded. But > I was wondering if any work has been done on the VoiceMail app to > actually store and retrieve messages to/from an object store? > > Cheers, > Andrew > -- > Andrew Ruthven, Wellington, New Zealand > MIITP, ITCP > > At work: andrew.ruth...@catalyst.net.nz > At home: and...@etc.gen.nz > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail using object storage?
Hey, I've found a bit of chatter about people using hacks to copy voicemail messages into object storage (like S3) after they've been recorded. But I was wondering if any work has been done on the VoiceMail app to actually store and retrieve messages to/from an object store? Cheers, Andrew -- Andrew Ruthven, Wellington, New Zealand MIITP, ITCP At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones
Richard Check both the DTMF settings, and the DialPlan string for account 3 on the phone. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Richard Schroeder" <rsch...@gmail.com> Sent: Tuesday, February 9, 2016 12:58 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones Perhaps this is not limited to Grandstream GXP 2000 phones, but those are the phones we are using. Using FreePBX. Retrieving a voice message (*97) works fine from Line 1. Retrieving a voice message (*98) and picking the extension (Comedian mail) works fine from Line 1. From Line 3, it does not recognize the password. (*97 or *98). The extension is installed on Line 3. Retrieving Line 3's voice messages can only be done from Line 1 (on any extension on the PBX). Line 3 seems to work fine otherwise. Is this a limitation, or is it some kind of setup issue? I can't seem to find anything in the documentation for the phone or FreePBX related to this issue. Anyone? This is frustrating and I will be grateful for any help. Thank you! Richard -- Richard C. Schroeder rsch...@gmail.com rsch...@optonline.net 516-859-1129 - Cell -- _ -- 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] Voicemail issue on Grandstream GXP2000 phones
: From Line 3, it does not recognize the password. Did you check whether you have the same DTMF settings for Line 3? -- _ -- 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] Voicemail issue on Grandstream GXP2000 phones
Perhaps this is not limited to Grandstream GXP 2000 phones, but those are the phones we are using. Using FreePBX. Retrieving a voice message (*97) works fine from Line 1. Retrieving a voice message (*98) and picking the extension (Comedian mail) works fine from Line 1. >From Line 3, it does not recognize the password. (*97 or *98). The extension is installed on Line 3. Retrieving Line 3's voice messages can only be done from Line 1 (on any extension on the PBX). Line 3 seems to work fine otherwise. Is this a limitation, or is it some kind of setup issue? I can't seem to find anything in the documentation for the phone or FreePBX related to this issue. Anyone? This is frustrating and I will be grateful for any help. Thank you! Richard -- Richard C. Schroeder rsch...@gmail.com rsch...@optonline.net 516-859-1129 - Cell -- _ -- 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] Voicemail: saycid without prefix
Nice! I didn't know what dialing rules may apply to his location, Your code does look like an improvement on mine tho. I love the REGEX function. Even better, if the first 4 digits are 0049, you could replace them with 0 as though it was an inland call: ExecIf(REGEX(^0049. ${CALLERID(NUM)})?Set(CALLERID(num)=0${CALLERID(NUM):4})) -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] Voicemail: saycid without prefix
On Monday 06 Jul 2015, Luca Bertoncello wrote: John Kiniston johnkinis...@gmail.com schrieb: The easiest solution may be to strip the leading zero's off your caller ID before your caller enters the Voicemail app to leave you a message. ExecIf(REGEX(^[0][0]. ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) Thanks! I already had this idea and implemented it. It works... Even better, if the first 4 digits are 0049, you could replace them with 0 as though it was an inland call: ExecIf(REGEX(^0049. ${CALLERID(NUM)})?Set(CALLERID(num)=0${CALLERID(NUM):4})) -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail: saycid without prefix
The easiest solution may be to strip the leading zero's off your caller ID before your caller enters the Voicemail app to leave you a message. ExecIf(REGEX(^[0][0]. ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello lucab...@lucabert.de wrote: Hi list! Yesterday I set up a voicemail on my Asterisk 1.8. It works as expected, but I'd like to have the CID without unnecessary prefix... Right now, if I call from my mobile phone I hear the complete prefix for my mobile number, indeed without 00. So I hear message from 49177 How can I set Asterisk to just read the prefix if it's necessary (so that calls from german numbers will not have 0049)? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] Voicemail: saycid without prefix
John Kiniston johnkinis...@gmail.com schrieb: The easiest solution may be to strip the leading zero's off your caller ID before your caller enters the Voicemail app to leave you a message. ExecIf(REGEX(^[0][0]. ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) Thanks! I already had this idea and implemented it. It works... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voicemail: saycid without prefix
Hi list! Yesterday I set up a voicemail on my Asterisk 1.8. It works as expected, but I'd like to have the CID without unnecessary prefix... Right now, if I call from my mobile phone I hear the complete prefix for my mobile number, indeed without 00. So I hear message from 49177 How can I set Asterisk to just read the prefix if it's necessary (so that calls from german numbers will not have 0049)? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voicemail ODBC Storage
On Saturday, October 25, 2014 09:09:57 PM Dan Journo wrote: Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? It may cause you trouble when using PostgreSQL: https://issues.asterisk.org/jira/browse/ASTERISK-24441 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Voicemail ODBC Storage
Hi, Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? Kind regards, Dan -- _ -- 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] Voicemail ODBC Storage
On Sat, Oct 25, 2014 at 4:09 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? Yes. app_voicemail uses a message envelope file to hold the metadata regarding the voice mail. When the ODBC retrieve function pulls the database records, it writes that data out to a temporary message envelope file for playback/manipulation by other functions. This process does not examine the column types, but instead simply looks at the column names and writes the data values out to the file using the types that it expects each column name to have. So, changing those types will not work out well for you. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail message number off by one when using ODBC storage
Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: Just wanted to let you know you were just left a 0:03 long message (number 7) but in attach there is the msg0006.wav Leandro -- _ -- 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] Voicemail message number off by one when using ODBC storage
... 'cause message file names start with 0 (msg.wav). -- marie On 05.10.2014, at 18:45, Leandro Dardini ldard...@gmail.com wrote: Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: Just wanted to let you know you were just left a 0:03 long message (number 7) but in attach there is the msg0006.wav Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Realtime
Hello everyone, I'm wondering if someone could help me. I would like to configure Voicemail users in realtime, by to not realtime voicemail storages. So i would like to have all voicemail accounts in database, but all voice message to be stored on disk. -- _ -- 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] Voicemail Realtime issue Failed to obtain database object for
Hello everyone, I would be extremely glad if someone could help me wilth my issue, here is are my configurations: odbc: http://pastebin.com/VPpfErYn mysql: asterisk sippeers http://pastebin.com/Y3vbSVda asterisk voicemail: http://pastebin.com/Ty3dbpGX Here are warnings from CLI: http://pastebin.com/D245r6xX Thank you, Best Ragards, Vadim -- _ -- 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] voicemail with odbc
Hi Rusty, Noted with thanks. Regards, Thet Tun On Thu, May 29, 2014 at 5:09 PM, Rusty Newton rnew...@digium.com wrote: On Thu, May 29, 2014 at 3:33 AM, ProNek pro...@gmail.com wrote: Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed snip You already started another mailing list thread on this topic a few hours before this. Please don't do that in the future. If you are going to post again, just post to the thread you already started instead of starting a new one. Did you double-check your database table carefully against the required schema? https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] voicemail with odbc
Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed *Error messages when I make call and leave message.* -- SIP/1ffa9-0007 Playing 'auth-thankyou.g722' (language 'en') [2014-05-28 14:55:13] DEBUG[12260][C-0006]: app_voicemail.c:3824 last_message_index: Directory '/var/spool/asterisk/voicemail/default/701/INBOX' has no messages and therefore no index was retrieved. == Parsing '/var/spool/asterisk/voicemail/default/701/INBOX/msg.txt': Found [2014-05-28 14:55:13] WARNING[12260][C-0006]: app_voicemail.c:4086 insert_data_cb: SQL Direct Execute failed! [2014-05-28 14:55:13] WARNING[12260][C-0006]: res_odbc.c:608 ast_odbc_direct_execute: SQL Execute error! Verifying connection to asterisk [asterisk-connector]... [2014-05-28 14:55:13] WARNING[12260][C-0006]: app_voicemail.c:4086 insert_data_cb: SQL Direct Execute failed! [2014-05-28 14:55:13] WARNING[12260][C-0006]: app_voicemail.c:4202 store_file: SQL Execute error! [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag,msg_id) VALUES (?,?,?,?,?,?,?,?,?,?,?,?)] == Parsing '/var/spool/asterisk/voicemail/default/701/INBOX/msg.txt': Found == Parsing '/var/spool/asterisk/voicemail/default/701/INBOX/msg.txt': Found -- Auto fallthrough, channel 'SIP/1ffa9-0007' status is 'NOANSWER' *Dialplan Configuration* [internal] exten = 701,1,Dial(SIP/ffbb,17,tT) same = n,VoiceMail(${EXTEN}@default,u) exten = 702,1,Dial(SIP/xlite-1,17,tT) same = n,VoiceMail(${EXTEN}@default,u) exten = 703,1,Dial(SIP/ffa9,17,tT) same = n,VoiceMail(${EXTEN}@default,u) *Voicemail Configuration* [general] format=wav49|wav attach=yes maxmsg=999 maxsecs=600! minsecs= 3 skipms=3000 maxlogins=3 odbcstorage=asterisk odbctable=voicemessages emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just ${IF($[${VM_CIDNUM} = ${ORIG_VM_CIDNUM}]?left:forwarded)} a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE},\n${IF($[${VM_CIDNUM} = ${ORIG_VM_CIDNUM}]?so:(originally sent by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n emaildateformat=%A, %B %d, %Y at %r pagerdateformat=%A, %B %d, %Y at %r tz=me ; Timezone from zonemessages below. Irrelevant if envelope=no. eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HMi me=Asia/Dubai|'vm-received' Q 'digit/at' H N 'hours' [default] ;1234 = 4242,Example Mailbox,root@localhost 701 = -7012,User One,pronek...@gmail.com 702 = -7023,Soft Phone,sp@localhost 703 = -7034,Pro Nek,pro...@gmail.com *res_odbc Configuration* [asterisk] enabled = yes dsn = asterisk-connector username = thet password = MyPassword pooling = 1 limit = 5 pre-connect = yes *func_odbc Configuration* [SQL] dsn=mysql1,asterisk readsql=${ARG1} ; ODBC_ANTIGF - A blacklist. [ANTIGF] dsn=mysql1,mysql2 ; Use mysql1 as the primary handle, but fall back to mysql2 ; if mysql1 is down. Supports up to 5 comma-separated ; DSNs. dsn may also be specified as readhandle and ; writehandle, if it is important to separate reads and ; writes to different databases. readsql=SELECT COUNT(*) FROM exgirlfriends WHERE callerid='${SQL_ESC(${ARG1})}' syntax=callerid synopsis=Check if a specified callerid is contained in the ex-gf database ; ODBC_PRESENCE - Retrieve and update presence [PRESENCE] dsn=mysql1 readsql=SELECT location FROM presence WHERE id='${SQL_ESC(${ARG1})}' writesql=UPDATE presence SET location='${SQL_ESC(${VAL1})}' WHERE id='${SQL_ESC(${ARG1})}' *voicemail show command* abox*CLI voicemail show users You must specify a specific context to show users from realtime! Usage: voicemail show users [for context] Lists all mailboxes currently set up abox*CLI *extconfig Configuration file* voicemail =mysql,asterisk,voicemessages I create table voicemessages in mysql exactly as description in Definitive Guide 4th edition book. -- _ -- 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] voicemail with odbc
On Thu, May 29, 2014 at 3:33 AM, ProNek pro...@gmail.com wrote: Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed snip You already started another mailing list thread on this topic a few hours before this. Please don't do that in the future. If you are going to post again, just post to the thread you already started instead of starting a new one. Did you double-check your database table carefully against the required schema? https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail message to text
Hi, we implemented ispeech for voice recognition. I works fine. But you have to develop an app of your own to do it. Take a look at http://www.ispeech.org/api (Section 3 Automated Speech Recognition). ispeech let you upload a recorded speex file via http-upload and will return the result at once as http-result. On their website you will find some code also to implement their service in any app. It's simple and you will get a quick result. Best regards -Thorsten- Am 20.05.2014 16:35, schrieb Ishfaq Malik: HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e:i...@pack-net.co.uk mailto:i...@pack-net.co.uk w:http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail message to text
HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail message to text
On 20 May 2014, at 15:35, Ishfaq Malik i...@pack-net.co.uk wrote: I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? With the huge variety of different accents and intonations in human speech (even in one country), my experience of all speech-to-text engines has been one of poor accuracy at best. If you need messages-to-text, generally best to use a virtual PA company or similar - at least in my experience. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] VoiceMail Issue
Any ideas on why this may not be working please ? - Original Message - From: Phil Daws ux...@splatnix.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 28 February, 2014 5:39:54 PM Subject: [asterisk-users] VoiceMail Issue Hello, am attempting again to resolve an issue with multi-tenancy and the forwarding to VMs between mailboxes. If in a multi-tenancy environment one uses custom contexts ie. [a1-ext1](a1) mailbox=101@a1 and the associated voicemail.conf entry: [a1] 101 = 1234,My User 1,ad...@email.com,,tz=eastern|imapuser=ad...@email.com|imapfolder=Inbox 102 = 1234,My User 2,ad...@email.com,,tz=eastern|imapuser=ad...@email.com|imapfolder=Inbox now if a message is left in mailbox 101 and the user attempts to forward the message to mailbox 102 Asterisk responds that mailbox 102 is not found in context default! One can add: searchcontexts=yes but that means each mailbox must have a unique number which goes against being able to use custom contexts. I thought by specifying the following would fix that: exten = 7999,1,VoiceMailMain(${CALLERID(num)}@a1) ; Direct mail retrieval exten = 7999,n,Hangup() but it does not. Have tried many ways to resolve but cannot find a resolution. Any ideas please as would like to get this working ? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail Issue
Hi Could you send us the logs from the asterisk? Carlos On Sat, Mar 8, 2014 at 4:03 AM, Phil Daws ux...@splatnix.net wrote: Any ideas on why this may not be working please ? - Original Message - From: Phil Daws ux...@splatnix.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 28 February, 2014 5:39:54 PM Subject: [asterisk-users] VoiceMail Issue Hello, am attempting again to resolve an issue with multi-tenancy and the forwarding to VMs between mailboxes. If in a multi-tenancy environment one uses custom contexts ie. [a1-ext1](a1) mailbox=101@a1 and the associated voicemail.conf entry: [a1] 101 = 1234,My User 1,ad...@email.com,,tz=eastern|imapuser=ad...@email.com |imapfolder=Inbox 102 = 1234,My User 2,ad...@email.com,,tz=eastern|imapuser=ad...@email.com |imapfolder=Inbox now if a message is left in mailbox 101 and the user attempts to forward the message to mailbox 102 Asterisk responds that mailbox 102 is not found in context default! One can add: searchcontexts=yes but that means each mailbox must have a unique number which goes against being able to use custom contexts. I thought by specifying the following would fix that: exten = 7999,1,VoiceMailMain(${CALLERID(num)}@a1) ; Direct mail retrieval exten = 7999,n,Hangup() but it does not. Have tried many ways to resolve but cannot find a resolution. Any ideas please as would like to get this working ? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail Issue
Hello, am attempting again to resolve an issue with multi-tenancy and the forwarding to VMs between mailboxes. If in a multi-tenancy environment one uses custom contexts ie. [a1-ext1](a1) mailbox=101@a1 and the associated voicemail.conf entry: [a1] 101 = 1234,My User 1,ad...@email.com,,tz=eastern|imapuser=ad...@email.com|imapfolder=Inbox 102 = 1234,My User 2,ad...@email.com,,tz=eastern|imapuser=ad...@email.com|imapfolder=Inbox now if a message is left in mailbox 101 and the user attempts to forward the message to mailbox 102 Asterisk responds that mailbox 102 is not found in context default! One can add: searchcontexts=yes but that means each mailbox must have a unique number which goes against being able to use custom contexts. I thought by specifying the following would fix that: exten = 7999,1,VoiceMailMain(${CALLERID(num)}@a1) ; Direct mail retrieval exten = 7999,n,Hangup() but it does not. Have tried many ways to resolve but cannot find a resolution. Any ideas please as would like to get this working ? Thank you. -- _ -- 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] Voicemail greeting playback issues?
From: Doug Lytle supp...@drdos.info Sent: Monday, November 25, 2013 6:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail greeting playback issues? Bryant Zimmerman wrote: Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange things can and will happen. Doug -- Doug The real issue here is that issuing an Answer() just before does not seem to solve the problem. To work around the issue I have to either put a Wait(1) or Dial() some extensions first. It is presenting like if you drop into the Voicemail() command too fast during call setup that you have issues. This did not occur in 1.8.x. I would be ok if just issuing an Answer() would resolve it as this would be normal, but having to slow down the dial plan seems off. Thanks Bryant -- _ -- 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] Voicemail greeting playback issues?
Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked and the greeting files are there and play back from the voicemail ivr. If no greeting is there it just plays The Pers.. beep Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. Any Ideas? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Voicemail greeting playback issues?
Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. I don't see this under 11.5.1 Doug -- _ -- 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] Voicemail greeting playback issues?
From: Doug Lytle supp...@drdos.info Sent: Monday, November 25, 2013 2:01 PM To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail greeting playback issues? Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. I don't see this under 11.5.1 Doug --- Doug Thank you for your response. It is good to hear that you are not having the issue. It gives me hope that there is a way to resolve this quickly. Do you have an thing special around your voicemail configuration? We started with the 11.xx sample config and mapped our settings from 1.8.x. Both our 11.xx and 1.8.x systems are running on the same virtual server. Both are reading and writing audio and vm files to and from the local storage. I forced off g729 to ensure that it was not causing the issues. Do you know of any way to force a higher level of debugging to see why the voicemail application would be having an issue? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Voicemail greeting playback issues?
From: Bryant Zimmerman brya...@zktech.com Sent: Monday, November 25, 2013 2:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail greeting playback issues? From: Doug Lytle supp...@drdos.info Sent: Monday, November 25, 2013 2:01 PM To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail greeting playback issues? Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. I don't see this under 11.5.1 Doug --- Doug Thank you for your response. It is good to hear that you are not having the issue. It gives me hope that there is a way to resolve this quickly. Do you have an thing special around your voicemail configuration? We started with the 11.xx sample config and mapped our settings from 1.8.x. Both our 11.xx and 1.8.x systems are running on the same virtual server. Both are reading and writing audio and vm files to and from the local storage. I forced off g729 to ensure that it was not causing the issues. Do you know of any way to force a higher level of debugging to see why the voicemail application would be having an issue? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. My voice mail test scripts do not answer or wait they just drop you into the voicemail box. It appears that something with Asterisk 11.xx is causing the voicemail() command to drop in and ether not play or mess up the prompts. If you have not given it at least one second in the channel before passing it to the voicemail() command. If you throw a wiat(1) just before the voicemail() command the prompts play correctly. So if you have rung extensions using dial() before going to voicemail that appears to be enough time. If you place an inbound call directly to voicemail() with no pause then you have an issue. Example Broken: exten = _9XXX,1,Set(l_VMExt=${EXTEN:1}) exten = _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext}) exten = _9XXX,n,GotoIf($[${VMBOXEXISTSSTATUS}=FAILED]?doHangup) exten = _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u) exten = _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup) exten = _9XXX,n,Hangup() Example Works: exten = _9XXX,1,Set(l_VMExt=${EXTEN:1}) exten = _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext}) exten = _9XXX,n,GotoIf($[${VMBOXEXISTSSTATUS}=FAILED]?doHangup) exten = _9XXX,n,Wait(1) exten = _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u) exten = _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup) exten = _9XXX,n,Hangup() The code that is broken with Asterisk 11.xx worked in Asterisk 1.8.x Can anyone confirm this? Thanks Bryant Zimmerman() -- _ -- 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] Voicemail greeting playback issues?
Bryant Zimmerman wrote: Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange things can and will happen. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Voicemail greeting playback issues?
On 11/26/2013 12:24 AM, Doug Lytle wrote: Bryant Zimmerman wrote: Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange things can and will happen. That's what I do since the 0.x days. IIRC in recent Asterisk versions some apps answer before doing anything else. Guess the voicemail app is not one of them. I always answer first followed by a small Wait and then execute the app. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail greeting playback issues?
On Mon, Nov 25, 2013 at 7:17 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 11/26/2013 12:24 AM, Doug Lytle wrote: Bryant Zimmerman wrote: Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange things can and will happen. That's what I do since the 0.x days. IIRC in recent Asterisk versions some apps answer before doing anything else. Guess the voicemail app is not one of them. I always answer first followed by a small Wait and then execute the app. VoiceMail does automatically Answer a channel. I'm going to guess that you have strictrtp enabled (which it is by default), and that if you cranked up Asterisk verbose logging to at least 4, you'd see something like this at about the time you started hearing audio: 0xYY - Probation passed - setting RTP source address to xxx.xxx.xxx.xxx Asterisk drops RTP packets until it locks onto an RTP source. It does this to prevent media injection attacks. The default probation period for an RTP source is four packets - you can configure the probationary period as well as whether or not strict RTP checking is enabled in rtp.conf. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail interface
Hello; Is there Interface (web based interface) that I can login as admin, check the emails and see the numbers that leaved voicemail and if possible to hear the voice message, ... etc? Regards Bilal-- _ -- 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] Voicemail Prepend Message Forwarding Not Working
Hi All, First I've heard of this feature not working from a customer. I did some digging and this is a common bug in several older Asterisk versions, it has more than a few patches in the bug tracker. I've tried a few of them but none will apply to a specific version I'm currently running for a customer, 1.6.0.28. Does anyone have a patch file that will apply to this version or an app_voicemail.c file that is already patched and will compile with this versions to fix this particular bug? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Voicemail Prepend Message Forwarding Not Working [SOLVED]
Hi All, First I've heard of this feature not working from a customer. I did some digging and this is a common bug in several older Asterisk versions, it has more than a few patches in the bug tracker. I've tried a few of them but none will apply to a specific version I'm currently running for a customer, 1.6.0.28. Does anyone have a patch file that will apply to this version or an app_voicemail.c file that is already patched and will compile with this versions to fix this particular bug? I patched app_voicemail.c manually from the patch file (revision 233691), recompiled and now prepending voicemail works. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Voicemail variables on email subject
RdSS == Rafael dos Santos Saraiva rafaels...@gmail.com writes: RdSS emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR} RdSS Return: RdSS Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?= That is a proper encoding for an SMTP mail header which is in utf8. RdSS Expected: RdSS Subject: 1504|12|Teste - Rafael 1570|16 The sent header decodes to this string: Subect: 1504|12|Teste_-_Rafael_1570|0:16 Note the colon from $VM_DUR (minutes:seconds). MUAs are supposed to decode that. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] Voicemail variables on email subject
I checked the raw text of my voicemail messages today and I saw pretty much the same escape sequences for UTF-8 like you did, but I do not have any display problem. You could save the message locally and hand edit it (starting with uppercase UTF instead of lowercase utf). jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail variables on email subject
I noticed that the problem occurs when I use the variables ${VM_DUR} and ${VM_CALLERID}. Only the subject of the message, if the body is not the problem. Using UTF or utf the same problem occurs. Att, *Rafael dos Santos Saraiva* Tel: (51) 8174-7956 *Digium Certified Asterisk Administrator (dCCA)* http://www.astdocs.com | http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 2013/8/6 jg webaccou...@jgoettgens.de I checked the raw text of my voicemail messages today and I saw pretty much the same escape sequences for UTF-8 like you did, but I do not have any display problem. You could save the message locally and hand edit it (starting with uppercase UTF instead of lowercase utf). jg -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail variables on email subject
I checked your original message, and I guess the expected string was a little bit different: 1504|12|Teste - Rafael 1570|0:16 I can't see anything wrong with quoted printable decoding. My best guess is still the email client and its settings. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users