search for: kvarsel

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. > > > > *Fr?n:* asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] *F?r *Patrick Wakano > *Skickat:* den 21 februari 2018 04:29 > *...

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

Hello, I have a broblem in asterisk 15 where an ami originate suddenly converts 58#+46435345534 to 58%23+46435345534. This happenend when upgrading asterisk 1.8 to 15. Could anyone help me with how to resolve this issue? Regards / Marcus [Beskrivning: Fogwise - logotype] Marcus Kvarsell phone: +46766350384 e-mail: marcus at fogwise.se url: http://www.fogwise.se Like us on facebook: https://www.facebook.com/WiseDialer<https://www.facebook.com/WiseDialer/app_362387440529737> Follow us on LinkedIn https://www.linkedin.com/company/fogwise-ab FOGWISE AB Fleminggatan 2 SE-602 24...

...gt meddelande----- Fr?n: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] F?r Joshua Colp Skickat: den 19 februari 2018 11:58 Till: asterisk-users at lists.digium.com ?mne: Re: [asterisk-users] # converts to %23 On Mon, Feb 19, 2018, at 4:24 AM, Marcus Kvarsell wrote: > Hello, > > I have a broblem in asterisk 15 where an ami originate suddenly > converts > 58#+46435345534 to 58%23+46435345534. This happenend when upgrading > asterisk 1.8 to 15. Could anyone help me with how to resolve this issue? You'll have to be more specific....

...serevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise - logotype] Marcus Kvarsell phone: +46766350384 e-mail: marcus at fogwise.se url: http://www.fogwise.se Like us on facebook: https://www.facebook.com/WiseDialer<https://www.facebook.com/WiseDialer/app_362387440529737> Follow us on LinkedIn https://www.linkedin.com/company/fogwise-ab FOGWISE AB Fleminggatan 2 SE-602 24...

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

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Im getting daily segfaults when running 40-100 cuncurrent calls in G729 passthrough mode. Any thoughs on why this is happening is most appreciated. #0 0x0000003cd773356f in __strlen_sse42 () from /lib64/libc.so.6 #1 0x000000000043b352 in update_bridgepeer (c0=0x7faeac06a320, c1=0x7faea4711570) at channel.c:4791 #2 0x0000000000445f42 in ast_channel_bridge (c0=0x7faeac06a320,