similar to: Macro arguments seperator

Hi! We have some problems with calls that get disconnected in the middle of a call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300, avgsilence 2090 Jan 9 14:56:17 DEBUG[4404] dsp.c: Requesting Hangup because the busy tone was detected on

The problem with the extra ptime descriptions in the SDP has been fixed in Asterisk (see http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html). I've got the latest version of the 1.4 branch from SVN and have verified that the codec negotiation is working again. If you don't want to try the latest SVN version, then you'll have to restrict the phones to a single codec

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

Hello! Is it legal to use RedirectAction to redirect a call that is waiting in a queue? The idea is to have an external application manage a queue via manager API. The queue would merely collect calls and play moh. I've tryed this already but asterisk sends SIP/Forbidden to the channel in queue, after the channel has been redirected by RedirectAction, even though the response to

I anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the firmware, read below; I installed one today (turnaround time around 2 weeks to Australia, inc. economy freight in both directions... impressive!) and have noticed immediate, significant improvements. Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking,

Hello We have discovered a problem with DTMF on Asterisk. We have a setup with a T1 from PSTN going into an Asterisk box, and then out again on T1 and into a normal PBX (EADS) We use it to record all calls going to/from the PBX. The problem is that when we record the calls (with MONITOR command), DTMF tones gets obscured, and is not understood in the other end, if we dont Monitor, there are no

Adding gluster-users. On Wed, Jan 31, 2018 at 3:55 PM, Misak Khachatryan <kmisak at gmail.com> wrote: > Hi, > > here is the output from virt3 - problematic host: > > [root at virt3 ~]# gluster volume status > Status of volume: data > Gluster process TCP Port RDMA Port Online > Pid >

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

hi, list R is undoudtedly my favorite statistic tool, however, the data inputnpart has long been a pain. most data I have to deal with are irregular and contains special character. Recently I get a tab delimited data, read.table(filename,sep="\t") constantly return erors for certain rows does not has xyz elements while all other programs such as perl,python, awk all report equal row

Ok, After following BJ's advice and removing ogg.so I then got a pbx_realtime.so error in the same fashion. I removed that file, and then the next and then the next as you can see in the log below. I think something is not right. duh here is my sign..lol...but I am not sure even where this ast_register_file_version flag is in a config file or what step I have missed. I am doing a VOIP only

Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL => (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41

Dear all after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt file) the | delimiter is not working by default. I've added a compat section in asterisk.conf a [options] dontwarn = yes [compat] pbx_realtime=1.4 res_agi=1.4 app_set=1.4 And restarted Asterisk, but i still have problem to have the | delimiter working, [Jun 9 23:20:54] DEBUG[11744]: pbx.c:3122

I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Johnathan Falk Network Administrator Clinton Community Schools

Hi, I have some interesting problem with htb , I set up root class and sub-classess: $TC qdisc add dev eth0 root handle 1: htb $TC class add dev eth0 parent 1: classid 1:1 htb rate 1990kbit ceil 2000kbit $TC class add dev eth0 parent 1:1 classid 1:10 htb rate 190kbit ceil 200kbit $TC class add dev eth0 parent 1:1 classid 1:11 htb rate 1400kbit ceil 1600kbit $TC class add dev eth0 parent 1:1

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]