similar to: Asterisk with RFC 3313 support

I have an Asterisk 1.2 install running on RedHat 9. I have a bunch of Polycom 501s co-locacted in the same building as *, and some more 501s in satellite offices (also registered to my * server) . Finally I have some road warriors running XLites. Ideally when a road warrior (XLite) calls a satellite office (Polycom 501), I'd like to avoid having Asterisk in the media path. I understand

I understand Asterisk is more like a B2BUA. But when this INFO request is sent to asterisk, asterisk is supposed to bridge the request to the other endpoint, right? In what situation, it decides to send a reply; in what situation, it decides to bridge the request? What is the role of gateway in SIP world, a proxy, a B2BUA or something else? Thank you, Wei Date: Fri, 18 Mar 2005 12:51:28 -0600

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP

When making an outbound call, the outbound peer return a 301 forwarded with URI to other domain, but asterisk think it's a local domain and try to look it up from extension.conf. How to configure so that a 301 forwarded with URI from other domain thinks it's outgoing to another proxy? thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the

We encouter a situation where we need to use SIP info to convey infomation for one end point to another endpoint. I use asterisk to do the test and find asterisk does not forward the SIP info to another endpoint, but act as UAS and returns a 4xx error message. I think asterisk is not right to handle this SIP info message. In RFC 3261 Page 70 "This protocol is designed to be extended.

Here's the bug and proposed patch. It's pretty trivial. https://bugzilla.mindrot.org/show_bug.cgi?id=1856 Quoting RFC-2474: A replacement header field, called the DS field, is defined, which is intended to supersede the existing definitions of the IPv4 TOS octet [RFC791] and the IPv6 Traffic Class octet [IPv6]. [...] The structure of the DS field shown above is incompatible with

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

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 ;

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

Hello, Is there an implementation for the RFC 4662 for asterisk 10? I found a patch for asterisk 1.8 but nothing for asterisk 10.12. The RFC: "This document presents an extension to the Session Initiation Protocol (SIP)-Specific Event Notification mechanism for subscribing to a homogeneous list of resources. Instead of sending a SUBSCRIBE for each resource individually, the

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

Is there a way to let chan_sip skip host lookup? Problem is I have to have a peer host config for every sip message outgoing. For example, I cann't have this in extension.conf exten => 500,n,Dial(SIP/romi at 192.168.1.79) It'll return, chan_sip.c:2738 create_addr: No such host: 192.168.1.79 when call forwarding I have to have a peer in SIP [outgoing] host=192.168.1.79 ... in

On Sun, Dec 21, 2014 at 4:54 AM, Recursive <lists at binarus.de> wrote: > Dear list, > > I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. > > 1) Ports and IP addresses which PJSIP bind to > > I have configured one transport

Bob, Have you looked at any of the products by Zyxel? With QOS, VPN & wireless support they have: For ADSL: Prestige 652HW Firewall/Router: Zywall 10W & 30W I'll be honest, I havn't used any of these yet. We were looking for similar products to suuport our VOIP installs. We just ordered some demo units from Zyxel, we shoul have them later this week. I'll let you know

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

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Hi! Sure, just look for: Wonder Shaper. It's a HTB based shaper configuration wich have some very good features, I use a variation of that here at my College. http://lartc.org/wondershaper/ It is the page (a simple google search). Also make sure to uncomment the line tos=lowdelay in every config file of asterisk that have it. Hope it is usefull, sincerely, Ildefonso Camargo

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

Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no "Proxy-Authorization" information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove