similar to: Updated: No audio when dialing in via PRI withQ.SIG

When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar. I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk

Thanks for you answer Marcus, So maybe this means some bug was fixed? Anyone aware of something related? >From the release notes, I couldn't find any direct change that could fix this.... Thanks, Kind regards, Patrick Wakano On 21 February 2018 at 20:29, Marcus Kvarsell <Marcus.Kvarsell at fogwise.se> wrote: > Hello, i found upgrading to asterisk 15 helped. > > > >

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Dear All, As an attempt to highlight the overall pattern in a Forest plot, I would like to "highlight" the area around HR=1. I cannot find any simple tools for painting a grey ribbon between 0.9 and 1.1. Any suggestions? Thank you in advance! Cheers, Patrik [[alternative HTML version deleted]]

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

Dear Davor We receive an error message at the command "list domains" ntdsutil metadata cleanup connections connect to server <DC with fsmo roles> quit select operation target error: error at handling the input invalid syntax -> list domains But the command is correct! Am 02.07.2015 um 21:11 schrieb Davor Vusir: > You might need to do a meta data cleanup before

Seems libvirt are broken in centos7? I can start it but Virsh list for example: Connection refused Kvm are installed And the kvm driver installed

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run this manually without any interruption in the system? And what does the script do? I

Hello, im using speex library on my vo ip project, but some times, after it works ok, it begins to play noises (like voice hits) decoding the packets of one source (the others sound ok). I finally found that reseting SpeexBits and decoder state it solves. If i cant avoid this problem, i wish to solve when it occours, but ?how to detect this? Please, if anyone has experiment this issue i

seems not possible to run a freebsd kvm guest on centos 6 all i get in the log using dmesg are something about mmio emulation failed -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos-virt/attachments/20130429/5036318e/attachment-0006.html>

Hallo When rejoining a dc to the domain I get the following error message: [2015/07/02 11:51:44.089786, 0] ../source4/librpc/rpc/dcerpc_util.c:729(dcerpc_pipe_auth_recv) Failed to bind to uuid e3514235-4b06-11d1-ab04-00c04fc2dcd2 for

:-) I don't insist on anything, I'm just struggling to learn a new language and partly a new way of thinking, and I really appreciate the corrections. I hope I someday will be able to handle lists in R as easy as I handle loops in Stata... Thanks again! Love -----Ursprungligt meddelande----- Fr?n: peter dalgaard [mailto:pdalgd at gmail.com] Skickat: den 4 december 2017 23:09 Till:

OK! I'm sorry about this. As I wrote earlier I'm totally lost... but I will try to explain the problem in steps bellow, ok. 1. I installed the rpm's for samba, Version 2.0.2a-ssl I think this is the version distributed with redhat linux 7.0 2. Then I changed the parameters in the /etc/samba/smb.conf file, and in this file I added the folowing parameters. [global] netbios name

I'm using speex under symbian (8000 hz, 16 bit) narrow band. The phones API only give me a buffer of 4096 bytes in recording.To reproduce audio I must fill up the buffer of the same dimension. 4096 isn't a multiple of 320. I want encode the audio in streaming. The solution that I adopt to encode is: - Divide 4096-256 bytes in 12 frames of 320 bytes. - Therefore the frame number 13 is

It's probably not practical to have them answering the client's telephone! At a lot of sites, incoming calls would be handled by auto attendant, diverted to answering service, etc. --Don -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian Nielsen Sent: Wednesday, May 10, 2017 2:46 PM To:

Hi again! I know you don't find loops evil (well, at least not diabolic :-) ). (After many hours googling I have realized that thinking about loops rather than lists is a newbie thing we Stata-users do, I just jokingly pointed it out). Anyway, I'm really happy that you try to teach me some R-manners. Since I still get questions about what the h**k I mean by my strange question, I sort it

Hello Asterisk list, I am facing some Asterisk crashes which are consistently pointing to the same backtrace, which is the following (using DONT_OPTIMIZE, BETTER_BACKTRACES and MALLOC_DEBUG): Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)): #0 0x00007f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6 #1 0x00000000004a91ca in cdr_object_get_by_name_cb () #2 0x0000000000463c60 in

Hi! Thanks for the replies! I understand people more accustomed to R doesn't like looping much, and that thinking about loops is something I do since I worked with Stata a lot. The syntax from Peter Dalgaard was really clever, and I learned a lot from it, even though it didn't solve my problem (I guess it wasn't very well explained). My problem was basically that I have a data matrix