similar to: RTP handling

On 03/24/2015 04:28 PM, Richard Mudgett wrote: > > > On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff at jeff.net > <mailto:jeff at jeff.net>> wrote: > > > Hello, > > I am wondering if asterisk does anything at all to RTP packets > passed from channel to channel if no transcoding is involved? Can > I assume that the packet that

On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff at jeff.net> wrote: > > Hello, > > I am wondering if asterisk does anything at all to RTP packets passed from > channel to channel if no transcoding is involved? Can I assume that the > packet that left phone A, arrived at the asterisk server, was copied to > phone B's channel and eventually arrived at phone B

Hello, http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly The link above indicates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing

Digital ocean offers ssd on all the virtual machines. Uptime is good. Jai Rangi Www.didforsale.com www.cebodtelecom.com www.cebod.com > On Mar 8, 2015, at 8:11 AM, Jeff LaCoursiere <jeff at jeff.net> wrote: > > > Amazon instances are shared resources. I wouldn't want to count on timing or disk throughput, and you can't just ask them to do "ssd" - its a

Hi Jeff Are you aware of any challenges of hosting it on AWS? It will help me to work out alternate plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance traffic. I see many posts referring to AWS deployment. Please help me to choose AWS server instance. *Thanks & Regards,*

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

I just got a spam from telephonydepot (which I invited to spam me, so I guess I have to call it legit marketing :) ), and they have some new device that is meant to be a surveillance camera with audio, but the interface is POE and SIP! A cool idea. Anyone playing with this toy yet? I am trying to wrap my head around how asterisk might fit with the model of this camera being

Hello, Did anyone succeeded in installing Asterisk on OpenWRT system. pls help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/aee7ee12/attachment.htm

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that

Has anyone here used T1s with RBS with asterisk? Cary Fitch

Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone

Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a "fax relay" setup locally with hylafax/iaxmodem to solve that issue,

Hello, I have a small LAN network connected through an Asterisk Server (Trixbox). I was looking to create my own custom made softphones, and I have been looking into how to transmit and receive via RTP. Would anyone know how the Asterisk RTP bridging works, and if there is any documentation on it? How is the RTP stream routed through the Asterisk server? Do I just give it the endpoints and

I am about to connect to a new provider who requires 20ms payload sizes in g729a. Is this configurable on asterisk? Is 20ms the default? Cheers, j

Howdy, I am playing around with asterisk within an LXC container on Ubuntu 11.04. I have asterisk (1.4.42) running fine, but want access to dahdi_dummy for timing (meetme). I have dahdi installed on the "host", and dahdi_dummy is loaded: root at astnorth:/# ls -ltr /dev/dahdi total 0 crw-rw---- 1 root root 196, 250 2011-06-08 13:59 transcode crw-rw---- 1 root root 196, 253

Hi, some questions/comments about performance/utility of * and * hardware I've been reading this list for a few weeks and I think I have compiled the better feelings of the users. please correct me if I'm wrong, still learning * .... Will be nice to see something like this in a wiki. After being flamed and corrected I will repost "clean" data. 1- Transcoding is the process of

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look

Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was "taken advantage of" over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN