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Anybody having issues with ztdummy under the current 2.6 RC7? I get the following errors when trying to modprobe ztdummy: "Unable to register zaptel rtc driver" Doing a Google on the error shows reference to a message from 2004 that said you might not have RTC compiled into the kernel. Checking via: cd /usr/src/linux-2.6.13-rc7 grep -i rtc .config shows: CONFIG_APM_RTC_IS_GMT=y

Hi, I'm trying to use the spandsp fax-back facility and despite I can send and receive faxes, this is not working fine. I would like to get a debug of what is going on. I am using the flag debug, but I don't know if txfax is generating any log information or where it is saving it. I just don't find anything. My line in extensions.conf is: exten =>

Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time.

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialp...

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > > > [from-siptrunk] > > exten => 13XXXXXXX,1,dial(${OPERATOR},20) > > > > exten => _13XXXXXXX,1,dial(${O...

I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

Hi, I use the latest Codeweavers WINE and several DOS programs wouldn't run correctly. Where can I find the best DOS emulator for Linux? Thanks!

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started

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

...of parkedcalls? This example works? [ramais] include => parkedcalls [parkedcalls] exten => 700,1,ParkedCall(701) exten => 702,1,ParkedCall(702) exten => 703,1,ParkedCall(703) exten => 704,1,ParkedCall(704) This exten works? 2016-02-03 17:27 GMT-02:00, Doug Lytle <support at drdos.info>: >>>> On Feb 3, 2016, at 2:19 PM, Vitor Mazuco vitor.mazuco at gmail.com wrote: > >>>> Humm, thanks for your reply >>>> But whats is the code in parkedcalls context. >>>> Please, can you give an example? > > > [ramais] > > i...

Humm, so this context parkedcalls is inside on features.conf? 2016-02-03 17:42 GMT-02:00, Doug Lytle <support at drdos.info>: >>>> On Feb 3, 2016, at 2:32 PM, Vitor Mazuco vitor.mazuco at gmail.com wrote: > >>>> Ah no, I'm asking what code I put inside of parkedcalls? > > Nothing, > > The context parkedcalls is generated by features.conf, you just need to > include i...

Thanks for trying, what asterisk version do you use? вт, 25 июн. 2019 г. в 17:50, Doug Lytle <support at drdos.info>: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. The > forwarded call showed my cell phone number, so I cannot reproduce. > > Doug &g...

...should keep the database of active registrations and unregister and reregister all of them during startup in case some of them remain active after unclean shutdown. Also probably provider side should be fixed? Thanks for your insight. Marek 2020-06-05 19:29 GMT+02:00, Doug Lytle <support at drdos.info>: > On 6/5/20 12:24 PM, Marek Greško wrote: >> How can this behavior been overriden? I do not expect this is problem >> on provider side, since it was working normally using chan_sip. > > Console output and dial plan snippets are always useful when diagnosing, > >...

I looked on the voip-info wiki and found sparse and conflicting information on how to do this with Asterisk... My incoming lines are all on Zaptel. Is there a simple why to implement a '*363" (do not disturb) toggle via the dialplan? It would be nice to be able to pick up an extension, dial *363, and have all calls sent to voicemail without ringing the extensions. Doing it again would

Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug

Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 -------------- next part -------------- An HTML