similar to: FollowMe reinvites

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Dear Martin, Does the work on nchar mean that bugs #16090 and #16091 will be resolved [1,2]? Thanks, Mark [1] https://bugs.r-project.org/bugzilla3/show_bug.cgi?id=16090 [2] https://bugs.r-project.org/bugzilla3/show_bug.cgi?id=16091 On Sat, Apr 25, 2015 at 11:06 PM, James Cloos <cloos at jhcloos.com> wrote: > >>>>> "GC" == G?bor Cs?rdi <csardi.gabor at

Does vorbis have any niches of technical superiority over opus? Or is compatibility with older hard- and software the only benfit? Put another way, is there any reason to prefer vorbis over opus for music on new sortware or platforms? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

>>>>> "DC" == D'Arcy Cain <darcy at VybeNetworks.com> writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> I disagree that it makes it worthless for a large number of JC> users. It's only within the last few days that a few people have run JC> into this particular issue where they have a public IP address that is JC> changing a lot and PJSIP does not support changing it without a JC> restart.

Trying to do a FollowMe test. When the extension is dialed, it dials my cellphone and my cell phone rings. But when I answer my cell phone it's just silence. When I press '1' on my cell phone, nothing happens. extensions.conf: exten => 140,1,FollowMe(mleonetti) followme.conf [general] featuredigittimeout=>5000 takecall=>1 declinecall=>2

The IANA has pre-allocated id 4 for ed25519. If waiting on the IANA were a reason to delay applying the SSHFP_KEY_ED25519 patch, it needn't be any longer. I've proposed un-reserving hash type 0 to be a "NULL hash", for those who'd rather publish the public key unhashed. Even if zero for unhashed fails to gain traction, I hope to see something allocated for that. But

Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly, will ast negotiate srtp or dtls even ast and the proxy speak sip in the clear over the lo interface? Avoiding encryption over lo can aid debugging, but will doing so also block secure media? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP:

It has been may years since I've done anything with a dahdi fxo; much has changed in the interim and I havne't found answers googling. The fxo hw is installed on the pots line in parallel to existing pots phones. My goal is to have a blf on the sip phone which lights whenever any of the devices on the pots line are off hook and which, when pressed, INVITEs the asterisk box such that it

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is

Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order If you need ast to register over tls, use something like this: register => tls://username:xxxxxx at sip-tls-proxy.example.org (copied from the

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which

Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel

Hi Adam, I appreciate your feedback, and understand totally where you're coming from as far as the database perspective goes. For the first "draft" of the app, I think I'm going to let it default to using a conf file for two reasons. First, my setup currently does not utilize a database. I would like to move to that type of a setup in the future however. Secondly, seeing as

Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call

Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES

Hello, I am writing an AGI script to achieve the following - Users can Disable/Enable followme from their extension. They can also change the followme details from their extensions. I have looked at the follow me table for freepbx. I can't see the field for the values enabling/disable followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL: