similar to: ISDN users: 1.6.x users, I need some testing done please, regarding Overlap Receiving

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

Copying to asterisk-users, as it's of use there too. I copied this code years ago from the net, it may have been modified since... This however is only used by managers, as it allows the manager to log a user in and out. For agent logged in/out status: where 8501 is the queue number and 8512 is the agent's extension, and SIP0001 is the agent's device. in extensions.conf

I've been asked for recommendations for a small call centre, an ethernet SIP deskphone with a wireless headset. Similar approach would be a mobile phone with bluetooth head set. Either I've not looked hard enough, or there isn't much on offer. Alec Davis -------------- next part -------------- An HTML attachment was scrubbed... URL:

What's with the occasional "Un-Top-posting", there is no rule that says you can't, http://www.asterisk.org/community/rules My preference is top posting, as you see the answer at a quick glance, instead of reaching for the scroll bar (or whatever key stokes are required) to get to the bottom, to find that the answer isn't there yet. Note: Flaming is not an acceptable

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Hi, I'm trying to replace function call with call to wrapper(function_name, num_args, ...), where varargs hold args of original call. Function* launch = Function::Create( TypeBuilder<int(const char*, int, ...), false>::get(context), GlobalValue::ExternalLinkage, "kernelgen_launch_", m2); { CallInst* call = dyn_cast<CallInst>(cast<Value>(I)); if

Hi Dimitry, This makes sense if you think about it from the perspective that the string you want passing must be passed at runtime, and so can't use a const char * from compile time. You need to make the string visible in the compiled image, and use that as the argument. A string is an array of 8-bit integers, so you need to create a ConstantArray. Value *v = ConstantArray::get(Context,

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT

Dear useRs, Some time ago I queried the list as to an efficient way of building a function which acts as ls() but with a different default for all.names: http://tolstoy.newcastle.edu.au/R/e6/help/09/03/7588.html I have struck upon a solution which so far has performed admirably. In particular, it uses ls() and not its explicit source code, so only has a dependency on its name and the name of

Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas

On 03/09/2020 23:25, Alec Moskvin wrote: > Hi Stephan, > > On Wednesday 02 September 2020 19:59:57, Stephan Bosch wrote: >> >> On 29/08/2020 21:04, Alec Moskvin wrote: >>> Hello, >>> >>> I have a rule to always delete a header. If the message gets fileinto'd, >>> the header is gone, but if it's delivered into the INBOX through

Hi to all.. I've setup a Samba domain and now having a hard time setting up Unix to Windows user mapping. As an example on the server, user is 'agi', and at the workstation I want an 'Alec Joseph' as the user name. If I log on from a Linux desktop using the alias connection goes through: # sudo tail -f /usr/local/samba/var/log.smbd | grep 'Alec Joseph' Got

Hi Stephan, On Wednesday 02 September 2020 19:59:57, Stephan Bosch wrote: > > > On 29/08/2020 21:04, Alec Moskvin wrote: > > Hello, > > > > I have a rule to always delete a header. If the message gets fileinto'd, > > the header is gone, but if it's delivered into the INBOX through > > implicit keep, the header does not get deleted. > > >

Since 2004 asterisk/libpri have been able to receive the Calling Sub Address in the ISDN setup message, and the dialplan was able to use it if required. It's support is limited to only NSAP, not BCD or user formatted. At the time 25/06/04 the questioned was asked, wouldn't it be a good idea to be able to transmit it as well, but that never got implemented, as it wasn't required at

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hi All, Just wondering is it safe to use asterisk 1.8 latest branch on production ? http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision 317100 -S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110505/373bd6fc/attachment.htm>

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

Will investigate.. On 28/08/2020 01:07, Alec Moskvin wrote: > Hello, > > I'm trying to set up the submission proxy, but if I set > submission_relay_ssl = starttls, it crashes. Without it, it works. > > Please find the details below. > > Thanks, > Alec > > > dovecot[256855]: submission-login: Login: user=<alec>, method=PLAIN, rip=::1, lip=::1,

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our