similar to: Alarms Sound files

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110509/cc00bd60/attachment.htm>

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

the schedule of my server this configured with -6:00, and this correct one with the normal hour of my country, I made the change but I don't work me [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include

Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in

Dear ALl Can we use Asterisk for only for transcoding?. if yes how many concurent call we can transcode with help of Astetrisk? For this we only need to config SIP.conf or any other file too. Thanks Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100704/f6159f70/attachment.htm

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071017/923a7b49/attachment.htm

Hey Rowland, I wanted to ask a. What is mutex.tdb, what is its purpose? b. why "mutex grab failed" started appearing in messages, what can be the probable reasons? Is pre-authentication failed related to locking failure of mutex.tdb? Dec 15 12:49:52 abc winbindd[30467]: [2018/12/15 12:49:52.764924, 0] ../source3/libsmb/cliconnect.c:1921(cli_session_setup_spnego_send) Dec 15

On Wed, 19 Dec 2018 12:32:45 +0530 Amit Kumar via samba <samba at lists.samba.org> wrote: > Hey, > > But What's the reason of this error? > I do not know, but what I do know is that, if it is a bug, you will never get it fixed in Samba 4.2.x, this version is EOL. It might also have been fixed in a later version. Sorry if this isn't what you want to hear. Rowland

Hi All, I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global default settings. but when user sending DTMf event with SIP info method my asterisk accepting that DTMF. If default or global setting is rfc2833 then how come asterisk accepting SIP info dtmf event? what to check please guide Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

About every hour I see the yellow alarms on all or a number of channels of my PRI which is connected to the dutch telephony network, Asterisk keeps on working fine.... Here's an example where channel 1-24 went into alarm: WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 1: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event):

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

For a long time--years-- I've been using Korganizer's Calendar. But lately-- since the last kernel upgrade, I believe-- the alarms don't work anymore. I set an appointment and (most of the time) schedule an alarm (one or more) for it, but the small windows containing the reminders don't pop up anymore... not even those set a year ago which always worked before. Anyone else

Greetings, I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover cable. I got it where the alarms are all ok/green but I'm unable to dial out or dial into the AS5200. Anyone have any suggestions as to where to begin troubleshooting this?

I have configured TDMoE sucessfully and I am able to make a Zap connection from one box to the other. The question I have is.. I'm getting repeated errors every second on both systems.. Oct 3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected alarm on channel 1: No Alarm Oct 3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected alarm on channel 2: No Alarm

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |