similar to: Fwd: Asterisk and COS bits

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 ;

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

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

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

Hello, i have a big problem... i want to connect my asterisk server to a lancom 1722 device (ISDN/SIP) Gateway. sip.conf: [general] context=default allowguest=yes realm=10.1.1.209 bindport=5060 bindaddr=0.0.0.0 tos_sip=cs3 ; f?r SIP-Pakete (Kommunikationsaufbau) tos_audio=ef ; f?r RTP-Audio-Pakete tos_video=af41 ; f?r RTP-Video-Pakete allow=all dtmfmode=rfc2833 canreinvite=yes

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

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

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

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

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

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Because it seems my mail from 30th august didn't make it to the list i send it again. If the mail _did_ get to the list and i didn't see it please excuse the duplicate post Below is the mail from the 30th: I have a setup like this: An asterisk-server with SIP-phones on the outside of a NAT. For example: asterisk with local IP-address

Hello, Several people report that outgoing phone calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context =

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 ~]#

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

Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker. Two SIP-Clients are connected (both on the local net, no NAT). The RTP stream flows directly between the phones. If I set phone A on hold, the music on hold is played. On the CLI I see the following message running: WARNING[2470]:

Hello, all. I'm having a deeply frustrating time getting multiple parking lots to work and am wondering what I am doing wrong. I am using Asterisk 1.6.1.1. I defined two separate parking lots in features.conf as follows: [parkinglot_a100] ; SSI context => a100-parking parkpos => 900-920 findslot => next [parkinglot_a10] ; EBC context => a10-parking parkpos => 800-820