similar to: Setting up RTP to flow between endpoints directly bypassing Asterisk

Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi -------------- next

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ?

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

I have a really simple question about a fairly complex problem: I have a Cisco 7960 behind a NAT. I have an Asterisk server behind a different NAT. I have a SER server (with rtpproxy installed) on a public IP adress. I've opened ports with static NAT to * and the Cisco. Without using SER, I can register the phone to *, I can complete calls, I just can't move audio. Reading the

Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven

Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route:

Hello, I wish to set a sip uri sip:info@mydomain. I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten => info,1,Answer() exten => info,n,Dial(Sip/84,10) exten => info,n,Dial(Sip/85,10) exten => info,n,Hangup Ser forward sip:info@mydomain

Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges between 10 to 60 ms delay but the average is near to 20 ms. We use SIP.

Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to

Hi asterisk, openser, ser users. I have to check video support between asterisk, open(ser) and rtpproxy . ASTERISK (b2bua+registrar server) | | | | SER + rtpproxy | | NAT | | sip agents (with video support) Both signalling and media channels are kept in the path of SER+rtpproxy and ASTERISK . I can

On Fri, Oct 12, 2018 at 02:41:59PM -0500, Bjorn Helgaas wrote: > s/iommu/IOMMU/ in subject > > On Fri, Oct 12, 2018 at 03:59:13PM +0100, Jean-Philippe Brucker wrote: > > Using the iommu-map binding, endpoints in a given PCI domain can be > > managed by different IOMMUs. Some virtual machines may allow a subset of > > endpoints to bypass the IOMMU. In some case the IOMMU

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

On 12/10/18 20:41, Bjorn Helgaas wrote: > s/iommu/IOMMU/ in subject > > On Fri, Oct 12, 2018 at 03:59:13PM +0100, Jean-Philippe Brucker wrote: >> Using the iommu-map binding, endpoints in a given PCI domain can be >> managed by different IOMMUs. Some virtual machines may allow a subset of >> endpoints to bypass the IOMMU. In some case the IOMMU itself is presented >

Hello List, I've searched the internet and I haven't been able to figure this out. I hope someone here can point me in the right direction. I have a MacOSX 10.3 machine and a WinXP machine that are being used to produce large video clips (@1gig). After they are rendered and ready for playback, they are copied to the playout machines using a system that uses watch folders shared out over

Dear R People: Hello again! I found something unusual in the behavior of the "endpoints" function from the xts package: > x1 <- ts(1:24,start=2008,freq=12) > dat <- seq(as.Date("2008/01/01"),length=24,by="months") > library(zoo);library(xts) > x2 <- zoo(1:24,order=dat) > #Here is the surprise: > endpoints(as.xts(x1),'quarters')