similar to: multicastRTP source interface

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Hello, I'm currently evaluating asterisk 13 (Currently on 11). We're testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Matt Hoskins wrote: > Good Morning, > > After further investigation, I found that the res_pjsip_publish_asterisk > module does not use the realtime sorcery wizard, but instead only reads > from the configuration files. I've been able to patch the module, using > the logic from the other modules to learn how to make the sorery > configuration read from the other sorcery

Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP. On each of those servers, there are a mix of SIP and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi from the voicemail server to the endpoint servers. Would this type of setup work

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 -------------- next part

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

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

Good Morning, After further investigation, I found that the res_pjsip_publish_asterisk module does not use the realtime sorcery wizard, but instead only reads from the configuration files. I've been able to patch the module, using the logic from the other modules to learn how to make the sorery configuration read from the other sorcery wizards and it's now working for the

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

Thanks we just figure it out a bit ago. It's amazing how simple some things are when you just ask - and then realized that you were making it too hard to begin with!! :-) Lane Hoskins, MCP Network Engineer 540.767.7626 -----Original Message----- From: Jared Smith [mailto:jsmith@drgutah.com] Sent: Tuesday, January 13, 2004 10:59 AM To: asterisk-users@lists.digium.com Subject: Re:

> 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 this, but Polycom uses their own form of MulticastRTP. It

Joshua, Regarding Outbound Registrations in realtime and a reload. Does it require a "pjsip reload" or full asterisk restart? Antonio G?mez Soto wrote: > Hi, > > The asterisk wiki page says: > > "Sorcery.conf allows you to try to configure other PJSIP objects such > as transport using realtime and it currently won't stop you from doing so. > However,

[this went to me instead of the list; MM, your list maintainer] I am digging in the wrong hole I guess. Where can I find R /S routines for spatial statistics? Also has anyone made an R link to a GIS package? Anyone out there who works in this area? Thanks Richard E. Hoskins WA State Department of Health 1102 Quince Street Olympia, WA 98504-7812 richard.hoskins at doh.wa.gov tel: (360) 236 -

Hello everyone, A month ago I took on an issue on the Asterisk issue tracker (https://issues.asterisk.org/view.php?id=11797) dealing with multicast RTP paging. This is the ability to send audio to phones (the phone must support it) and have it played out the speakerphone. Using multicast RTP is great for this because it does not incur the cost and weight of setting up a potentially short call.

To try and work through the initial drops, I repeated the previous steps that worked on the physical machine on the CentOS 7.6 virtual machine. The drops continued to occur, but it seems that the driver always reconnects. My initial inclination is to work through and see at what point I get the drops to stop, and I think I will do that next (inputs welcome). From a standpoint of “how much does

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 scrubbed... URL:

HI Guys, I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? My goal is to do the paging using a multicast IP address. I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast. https://issues.asterisk.org/view.php?id=11797 http://svnview.digium.com/svn/asterisk?revision=101218&view=revision My dialplan

Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the