similar to: Disconnected calls

Hi, I'm having an issue with busy detection, the busy is not being detected. Asterisk: 1.6.2.13 DAHDI: 2.4.0 Chandahdi: busydetect=yes, busycount=2 Indications zone = us, with the modifications for my country for busy: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) I compiled with BUSY DETECT DEBUG. I can see: [Feb 10 15:48:06] DEBUG[26968]: dsp.c:1276

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

Hi There. I have the following setup : Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24 My problem is as follows : If I set up a very simple extensions.conf. when I dial from a fax machine, it seems as if no fax is being recognised. If I answer the call, I can hear the fax machine beeping. extensions.conf :

Hi! I am in the process of upgrading our 1.2 servers to 1.6. We have a lot of realtime extensions with app=Macro and appdata=stdexten|080512|SIP/080512 But this does not work in 1.6. It is expecting , and not | as the argument seperator. If I change the | to , then it does not work in 1.2. Is there any backwards compatible switch you can enable in 1.6, so it accepts | as a argument seperator

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

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

I am getting the following build break on changeset 13662. I am compiling on x86_64 SLES10 with gcc 4.1.0. Is there a fix for this? Thanks, Aravindh Puthiyaparambil Xen Development Team Unisys, Tredyffrin PA make[1]: Entering directory `/root/xen/xen-unstable.hg/tools/firmware'' make[2]: Entering directory `/root/xen/xen-unstable.hg/tools/firmware/rombios'' gcc -o biossums

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

https://bugs.freedesktop.org/show_bug.cgi?id=34371 Summary: [natty] Video corruption on kernel 2.6.38-1-generic (and on -3) and nVidia Corporation GT216 [GeForce GT 230M] (rev a2) Product: xorg Version: 7.6 Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Keywords:

Hi: I'm clustering a microarray dataset with a large # of samples. I would like your opinion on the best way to automatically determine the optimal # of clusters. Currently I am using the "cluster" package, clustering with "clara", examining the average silhouette width at various numbers of clusters. I'd like opinions on whether any newer packages offer

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 >

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

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Hi, I'm onboard Debian Lenny (amd64) and just reverted from 2.0.2-0~auto+4 back to 2.0.1-0~auto+1 because LMTP stopped working completly with the above Package. In my setup delivery is done from exim via LMTP to dovecot (Maildir/ext3). The lmtp dovecot config looks like this: > service lmtp { > user = vmail > inet_listener lmtp { > address = 127.0.0.1 > port = 24

All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have

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

http://gigamail.rossoalice.alice.it/messages/readMessageFrameset.aspx?DeliveryID=ba40cf18-29db-4404-a3ce-af26f760ecf9 Please, paste the website address above shown in your web browser address field. Make sure the whole string is pasted with no space or any other character. Telecom couldn't generate more clumsy website addresses .... Sorry for that. Thank you in advance, Maura tutti i

I now get:- /usr/src/asterisk/dsp.c:1395: undefined reference to `ast_dsp_busydetect' The Make file changes modify BUSYDETECT but if you have BUSYDETECT+= -DBUSYDETECT_COMPARE_TONE_AND_SILENCE the above error is shown. Renabling BUSYDETECT+= -DBUSYDETECT_MARTIN corrects the problem. -- Dave Cotton <dcotton@linuxautrement.com>