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...en I add a context and an extension entry in voicemail.conf, it works the way it should. Is there something that I'm missing here? Any insight at all would be greatly appreciated. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) <mailto:fsd at voipbusiness.us> support at voipbusiness.us -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140730/b15f540c/attachment.html>

...ransport.o] Error 1 Makefile:402: recipe for target 'res' failed make: *** [res] Error 2 Has anyone seen this error before? Any insight at all would be greatly appreciated. Thanks; John V. Tech Support Tech Support VoIP Business Solutions 240-215-3479 x325 <mailto:fsd at voipbusiness.us> support at voipbusiness.us -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170418/ee3001df/attachment.html>

...Asterisk-14-on-Ubuntu.md Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit. I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16 and 17 so this should work! Let me know how you get on. On 18 April 2017 at 13:41, Tech Support <asterisk at voipbusiness.us> wrote: > All; > > I am trying to build and install certified Asterisk 13.13 cert3 on > a Ubuntu 16.04.2 LTS host without much success. I am getting the > following errors when I try to compile. > > > > [CC] res_pjsip/config_transport.c -> res_pjsip/...

...he two, so that writes can go to either server using only a single IP address configured in Asterisk. Then, if one fails, you can still write to (and read from) the other, repair the failed one, and restore replication. Antony > > On Jun 19, 2017, at 17:47, Tech Support <asterisk at voipbusiness.us> wrote: > > > > All; > > > > I know that there are probably several solutions to this problem, but > > what I am trying to do is provide some redundancy for my customers > > CDR data. I know that doing simple backups of MySQL is probably t...

...ax, jump to the fax extension. now when should i use these show commands to detect fax and how should i tell asterisk to execute fax extension? this is a big problem for me, so i really appreciate if you help me to solve it. yours, SAM On Wed, May 20, 2015 at 7:11 PM, Tech Support <asterisk at voipbusiness.us> wrote: > Hey; > > Yes, I?ve also seen that 5 second delay with our fax server and it > drove me crazy. How I solved it was by doing a ?core show channels > <concise|verbose>? and detect if there was a fax transmission going on. > Doing it this way shows up in...

...about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt > On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote: > > That's the basics, but you have to nail the timing just right. The timing is > really important to do it the right way. > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists....

On 04/29/2017 10:57 AM, Jonathan H wrote: > On 29 April 2017 at 16:47, Tech Support <asterisk at voipbusiness.us> wrote: > >> I?m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. However, when I try to compile it, I?m getting hundreds and hundreds of errors. Here is a sample of the output. >> When I try to build Asterisk 13, I have no problem. Any insight at all...

2015-05-06 17:51 GMT+02:00 Tech Support <asterisk at voipbusiness.us>: > Hey; > > It seems to me that for what you want to do, it would be easier just > to email the user the voicemail audio file as an attachment. > Yes that's true but still, for users with an "intensive voicemail usage", unified messaging is attractive....

Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support <asterisk at voipbusiness.us> wrote: > Please keep the ?me to? emails off the list. > > Regards; > > JV > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *Magno Guimar?es > *Sent:* Monday, June 22, 2015 3:5...

If I correctly understand what the problem is, what I did was write a script that runs out of CRON every 15 minutes. It checks the outside IP address by querying http://checkip.dyndns.org and compares it to the IP address stored in the parameter ?externip? in the [general] section of sip.conf. If the two values are the same, the script exits quietly. If they are different, the script updates

hello everybody i want to send fax via asterisk in pass through mode. everything is ok if enable fax detection in ooh323 and write fax extension in extensions.conf file. just one problem: delay. i have to wait 5 seconds in order to fax detection done. it is too long for me when i have voice call and no fax. my phone rings after five seconds. is there any way to omit or reduce this time? i test

...the remote end was a fax machine or a plain old phone. I would also like to determine that in the dial plan if I could. Any insight at all with this would be extremely helpful. Thanks; John V. Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) <mailto:fsd at voipbusiness.us> support at voipbusiness.us -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141208/77954264/attachment.html>

John, thank you four your answer. I think you have misunderstood the problem. It?s about a ip address change of the sip trunk, not of my asterisk server. Kind regards, Daniel > Am 01.04.2015 um 16:40 schrieb Tech Support <asterisk at voipbusiness.us <mailto:asterisk at voipbusiness.us>>: > > If I correctly understand what the problem is, what I did was write a script that runs out of CRON every 15 minutes. It checks the outside IP address by querying http://checkip.dyndns.org <http://checkip.dyndns.org/> and co...

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

All; I am running Asterisk 11.6-cert13 and am having a problem when parking a caller. A call comes in and I put them on hold with *1 (defined in features.conf) without a problem. I can then dial the parked call extension number, say 701, and retrieve the call. The problem I'm having is if I try to park the call again, nothing happens. I hear a beep when I hit *1, but thats it. Has

...ation in the CDR, but the "fax show stats" command only gives a summary since the last restart. It would be nice if the FAXOPT variables contained this information but it doesn't. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) support at voipbusiness.us <mailto:fsd at voipbusiness.us> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131014/1482a166/attachment.html>

...T38FaxUdpEC = t38UDPFEC udptlchecksums = no sip.conf: faxdetect=yes t38pt_rtp=no t38pt_tcp=no t38pt_udptl=yes,fec t38pt_usertpsource=yes Any help at all would be greatly appreciated. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 <mailto:fsd at voipbusiness.us> support at voipbusiness.us -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140121/4072bfae/attachment.html>

...someone point me in the right direction as far as documentation and examples go? I would greatly appreciate it and will make it all available publically if the implementation turns out well. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) support at voipbusiness.us <mailto:fsd at voipbusiness.us> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140128/5d6ec109/attachment.html>

All; I have an application that dials a list of numbers and then plays a recorded message. My customer uses it to dial a list of customers to confirm their appointment for the next day. No biggie, maybe 25 - 30 calls per day for customers who want the confirmation call. What they need now is a way to dial an extension after the number is dialed and answered. I've seen that before, but I

...as this number is owned by another carrier that we do not have a business relationship with." So my question is this. How do I find out which carrier owns the DID in question? Thanks; John V. Tech Support Tech Support VoIP Business Solutions 240-215-3479 x325 <mailto:fsd at voipbusiness.us> support at voipbusiness.us -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171102/eb88185f/attachment.html>