similar to: no silk translation ?

Hi After upgrading from 1.4 to 11.4.0, I am able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

Hallo again I now succeeded in using dbApply on my data and I can convert it into a data.frame. But as Peter Dalgaard pointed out I his answer to my earlier question (Re: [R] list to data frame, 17.07.2003) I get one row and 10000 columns instead of what I want two columns and 10000 rows when I convert the list that dbApply returns to a Data frame. The list I want to convert looks like this

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

Hi, I have just published the module I use to manage POSIX ACLs : fooacl I don''t consider it the cleanest possible approach to the problem, but it''s very efficient and flexible. I would actually call it a hack :-) There''s room for improvement, such as splitting out Execs per managed path to avoid useless re-applying on unchanged paths, or using file snippets without

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @

Hi After upgrading from 1.4 to 11.4.0, I *am* able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan <sonny.rajagopalan at gmail.com> wrote: > George, > > I have the detailed log below. (Resending after trimming the email to 40KB.) > > The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? > > Thanks! > I don't see anything obvious. The registration works though, right? You might want to compare

Any suggestion.... On Dec 6, 2017 11:51, "ABHISHEK PALIWAL" <abhishpaliwal at gmail.com> wrote: > Hi Team, > > We are getting the crash in glusterd after start of it. When I tried to > debug in brick logs we are getting below errors: > > [2017-12-01 14:10:14.684122] E [MSGID: 100018] > [glusterfsd.c:1960:glusterfs_pidfile_update] 0-glusterfsd: pidfile >

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my

Without the glusterd log file and the core file or the backtrace I can't comment anything. On Wed, Dec 6, 2017 at 3:09 PM, ABHISHEK PALIWAL <abhishpaliwal at gmail.com> wrote: > Any suggestion.... > > On Dec 6, 2017 11:51, "ABHISHEK PALIWAL" <abhishpaliwal at gmail.com> wrote: > >> Hi Team, >> >> We are getting the crash in glusterd after

On 06/03/16 17:28, Harry Jede wrote: > On 18:11:20 wrote Rowland penny: >> On 06/03/16 16:22, Harry Jede wrote: >>> On 17:13:33 wrote Martin Juhl: >>>> Hi guys >>>> >>>> >>>> When trying to set root's password, I get a segmentation fault: >>>> >>>> [root at bart ~]# smbpasswd -a root >>>> No

On 19:47:03 wrote Rowland penny: > > I have just started an old vm with samba 3.6.6 as pdc and openlap > > as backend. smbpasswd -a someuser does not work, if someuser does > > not exist. > > Are you using smbldap-tools or ldapsam:editposix ? In this vm ldapsam:editposix. OK. I have just created a posix-only user in openldap. And then tried smbpasswd -a test01.

Hi Atin, Please find the backtrace and logs files attached here. Also, below are the BT from core. (gdb) bt #0 0x00003fff8834b898 in __GI_raise (sig=<optimized out>) at ../sysdeps/unix/sysv/linux/raise.c:55 #1 0x00003fff88350fd0 in __GI_abort () at abort.c:89 [**ALERT: The abort() might not be exactly invoked from the following function line. If the trail function

I hope these logs were sufficient... please let me know if you require more logs. On Wed, Dec 6, 2017 at 3:26 PM, ABHISHEK PALIWAL <abhishpaliwal at gmail.com> wrote: > Hi Atin, > > Please find the backtrace and logs files attached here. > > Also, below are the BT from core. > > (gdb) bt > > #0 0x00003fff8834b898 in __GI_raise (sig=<optimized out>) at >

Good day. 1. We have configured two domain controllers on Windows 2003 R2. We named them TEST.LOCAL and CHILD.TEST.LOCAL respectively and made a trust relationships between them. 2. We have installed Samba 3.5.3 on Ubuntu 9.10, kernel 2.6.31-14 and configured it for using winbindd. We have encountered a problem with results that winbind returns upon a command GETGRENT. We

George, I have the detailed log below. (Resending after trimming the email to 40KB.) The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? Thanks! --------------------- Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 ---> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0 Via: SIP/2.0/UDP 18.18.19.123:5060

ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under