similar to: Asterisk and COS bits

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

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [170 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [170 at internal:2] NoOp("DAHDI/1-1", "Context: office-extensions") in new stack

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [Aug 2 20:27:50] == Using SIP VIDEO CoS mark 6 [Aug 2 20:27:50] ==

Hi, examining the network traffic with wireshark shows that asterisk does not set any QoS values at all. What do I need to do to make asterisk set QoS values (on Centos 7)? The wiki says to use vconfig to set QoS values[1]. What does the skb-priority need to be set to? How do you use vconfig on interfaces that are not VLAN interfaces? Is it generally impossible to set QoS values on

Well, correct me if I'm wrong, but I would say this conversation you have posted is a bit outdated, now fail2ban can be used with asterisk security log https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger. On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <support at telium.ca> wrote: > Keep in mind that the attacks you are seeing in the log are ONLY the

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Thanks, Michael. A few questions: Is [transport_name] a reserved word, or am I supposed to replace it with a name of my own, like '[did-transport]'? Some of the keywords I haven't seen before. Is ca_list_file supposed to be an aggregate of the public and private key? And what are the 'method,' 'tos' and 'cos' keywords, which are commented out in your

On Fri, Jan 31, 2020 at 7:34 AM hw <hw at gc-24.de> wrote: > Hi, > > examining the network traffic with wireshark shows that asterisk does not > set > any QoS values at all. > > What do I need to do to make asterisk set QoS values (on Centos 7)? > > The wiki says to use vconfig to set QoS values[1]. What does the > skb-priority > need to be set to? How do

Hello, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603 at DLPN_AlDimnaDialPlan:601] Dial("SIP/601-00000002", "SIP/603") in new stack [Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433 dial_exec_full: Unable to

Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter

We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722),

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( But when I used Vanilla Asterisk then All things are working.... Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf*

Hello Steve, use the following configuration for the transport and bind this transport to the trunk: [transport_name] type=transport protocol=tls bind=192.168.13.24 ; your bind IP ca_list_file=/etc/pki/tls/certs/ca-bundle.crt ; method=tlsv1_2 verify_server=yes allow_reload=no ;tos=0xb8 ;cos=3 external_media_address=your.ext.host.name ; hostname pointing to your ext. IP

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hi, With Asterisk 1.8.3, I can't figure out how to pass an argument to a macro which is used within an originate command. Here is my sample dialplan to illustrate: exten => 123,1,Answer() exten => 123,n,Originate(SIP/20,app,Macro,foo,bar) exten => 123,n,NoOp(This is the NoOp after the originate command) exten => 123,n,Wait(30) exten => 123,n,Hangup() [macro-foo] exten =>

Hi, i try sin, cos, and tan. > sapply(c(cos,sin,tan),function(x,y)x(y),1.23e45*pi) [1] 0.5444181 0.8388140 1.5407532 However, *pi results the following > sapply(c(cospi,sinpi,tanpi),function(x,y)x(y),1.23e45) [1] 1 0 0 Please try whether the following becomes all right. diff -ruN R-3.3.2.orig/src/nmath/cospi.c R-3.3.2/src/nmath/cospi.c --- R-3.3.2.orig/src/nmath/cospi.c 2016-09-15

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No