search for: multicastrtp

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data from MP3Player, I get no s...

I am using a cyberdata "sip paging adapter" and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... Asterisk 11.11.0 is what I am using. What might be wrong here? Thanks, jerry ----------...

Hi everyone, I am sending out a multicast page using the following in my dialplan: Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q) Everything works great, but I had a question about SIP and SDP: Should I be seeing a SIP/SDP message from the asterisk server containing media information and the multicast IP address? On wireshark, I see SIP/SDP from the admin phon...

I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. I'd like to use the multicastRTP driver to do paging. However, when a phone dials an extension with multicastRTP, the multicast stream goes to the primary interface (eth0) and it really needs to go to eth1. Is there a way to specify which interface the rtp is sourced from? Matt Hoskins | NPG Corp | Systems Architect 81...

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called MulticastRTP/basic/x.x.x.x:5555 -- MulticastRTP/0x7f8b4000f898 answered SIP/XXXXXXX-0000004c After connecting and hearing the "beep" the line stays open and I can talk and press buttons and so on, but the...

I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine. Is there something special to do to get multicast working with polycom phones? (other than enable multicast on the actual phone). Thanks Jerry -------------- next part -------------- An HTML attachment was...

> I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with > polycom phones as other devices receive my multicast just fine. > > Is there something special to do to get multicast working with polycom phones? > (other than enable multicast on the actual phone). Didn't see if anyone had answered you or not on thi...

...ng list who is interested and has the time to test to please test and provide some feedback. A branch based off of trunk (as that is where the channel driver will go) is available at http://svn.asterisk.org/svn/asterisk/team/file/issue11797 The dial string for the channel driver is in the form of MulticastRTP/<type>/<destination>/<control address> where type is either basic or linksys. The control address is only needed for the linksys type. Any feedback is welcome as a note on https://issues.asterisk.org/view.php?id=11797 and will help to getting this into the tree. Thanks! -- Jos...

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-Answer: normal); Any other ideas? Leandro PS I have set

...time to test to > please test > and provide some feedback. > > A branch based off of trunk (as that is where the channel driver will go) > is available at http://svn.asterisk.org/svn/asterisk/team/file/issue11797 > > The dial string for the channel driver is in the form of > MulticastRTP/<type>/<destination>/<control address> where type is either > basic or linksys. The > control address is only needed for the linksys type. > > Any feedback is welcome as a note on > https://issues.asterisk.org/view.php?id=11797 and will help to getting > this int...

...loading codecs g729/g723 (Reported by Alexei Gradinari) * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph) * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13 (Reported by Daniel Denson) * ASTERISK-26326 - Crash when dialing MulticastRTP channel (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell) * ASTERISK-22131 - Update the make dependencies script to pull, build, and ins...

I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm