similar to: Lancom 1722 and Asterisk (i need HELP)

My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

Hello. I'm using Asterisk 1.4.22 with Postgres 8.3 in a Ubuntu 8.04 Server. I configured Asterisk to get sip from Postgres, and set qualify for all sips as yes, but the sip show peers command show the status of the peers as UNKNOWN srvcentral*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status Realtime 4900/4900 (Unspecified) D

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

Hi, I am trying to fix a problem I have for about a week now. The environment is: --several independent servers with Centos5.4 on the latest patchlevel (example will be server2) All of them are working properly. --a machine with Xen installed to host some virtual machines (xenserver1) --domu servers on xenserver1 (crmserver1, winserver1) The whole network is connected with a Lancom router to a

Hi List, i have a realy strange problem with no solution yet, i''m using iproute together with the iptables mangle option, in a dmz network is a cisco pix present with another inet link behind, therefore i''m using the mangle option to split traffic on a protocol base like: iptables -A PREROUTING -t mangle -i eth1 -s 192.168.1.5 -p tcp --dport 80 -j MARK --set-mark 3 and add the

Hi, I''ve written a small javascript ADSL throughput calculator: http://nukunuku.yamamaya.is-a-geek.org/~ranma/adsl.html Feel free to submit alternative presets (I currently have presets for three german telecom speed variants: T-DSL (1000|2000|3000), derived from http://www2.lancom.de/kb.nsf/5d445c701b3ff52dc1256e7700297e5c/27c6ee1c3e3f74b0c1256e94004a433e?OpenDocument). Comments,

Hi, I tried to install packages for R, but it did not work. I looked for a hint on the R website, but could not find anything. I tried the following commands: install.packages("psych") install.packages("QuantPsyc") install.packages("car") install.packages("Hmisc") install.packages("foreign") install.packages("Rcmdr")

Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer

You have it right, for 1.2, use 'tos=', for 1.4 use 'tos_sip/tos_audio/tos_video'. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Al lists Sent: Monday, July 23, 2007 10:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fwd: Asterisk and COS

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

With a little delay, here's the announcement for out 3.0.24 packages. English version below. Samba 3.0.24 ist soeben ver?ffentlicht worden. RPM-Pakete f?r diverse SUSE und RedHat-Versionen sowie f?r Debian GNU/Linux k?nnen von http://ftp.sernet.de/pub/samba/ heruntergeladen werden. Pakete f?r S390 folgen in K?rze. Dieses Samba-Release behebt einige Fehler, darunter: * einen

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi Michael. I fear I'm becoming a pain in your back. Sorry. The problem with the "Components" and "Suite" name in the "InRelease" file for bionic-cran35 has been solved. *Many thanks for that.* However, "sudo apt-get install r-base" creates an error message (again here the German version): ulrich at linuxdesktop:~$ sudo apt-get install r-base

Am 02.12.19 um 11:43 schrieb Rowland penny via samba: > On 02/12/2019 10:07, Stefan G. Weichinger via samba wrote: >> Am 02.12.19 um 08:47 schrieb L.P.H. van Belle via samba: >>> Hai, >>> >>> Sorry for the late reply. >> Never mind, weekend is important ... >> >>> Here its just apt-get distupgrade --autoremove --purge >>> Without

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

In that case. apt-get update apt-get remove samba winbind --autoremove apt-get dist-upgrade --autoremove apt-get install samba winbind libnss-winbind libpam-winbind tdb-tools apt-get dist-upgrade --autoremove --purge As long as you dont use --purge, everything of samba will stay on the system and you able to install the buster version. Greetz, Louis > -----Oorspronkelijk bericht-----