similar to: Skinny directmedia

Is there a work around for Caller ID information not being persisted when using the CLI or AMI Channel Redirect. A calls B (caller id is displayed), B transfers call to C (no caller id is displayed on phone c). Jacob Miles Software Engineer jacob.e.miles at l-3com.com 903.457.4422 -------------- next part -------------- An HTML attachment was scrubbed... URL:

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

Asterisk Project Security Advisory - ASA-2007-016 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Remote crash vulnerability in Skinny channel | | | driver

Asterisk Project Security Advisory - ASA-2007-016 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Remote crash vulnerability in Skinny channel | | | driver

I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the "make linux install" command. I can execute lua scripts via the command line, but asterisk configure script is unable to

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

Please forgive me my ignorance ... I've spent two days trying to find out something about the format of the default configuration file, which CCM produces. The only example I have so far is the one from the chan_sccp source. There were tons of references on entering the callmanager commands on a cisco command line - which I don't have (don't need thanks to chan_skinny + chan_sccp).

Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the

At the present time you have to have a VALID ip address in bindaddr for Skinny to work. If bindaddr is either 0.0.0.0 or simply commented out all packets requiring the IP address contain 127.0.0.1. I forgot their nick, but someone in IRC recommended we make Asterisk be smart enough not to pick that interface, but I'm not sure of that is the problem or not. I simply have not had the

Ok, Skinny gurus. (btw, I'm super pleased to see development happen on this). Thoughts on this?? I added this context to my skinny.conf: [ppc] device=SEP00022D494F2A context=employees line => 50 ; Dial(Skinny/1@ppc) I've downloaded the 30 day Window eval of VTGO! (PPC) from www.ipblue.com and it hangs on Registering. *CLI> skinny debug Skinny Debugging Enabled

Hi, please tell me, is original skinny support in Asterisk stil under development or is better to try chan_sccp from http://chan-sccp.sourceforge.net ? my first try was unsuccessfull (chan_sccp compile OK, but module loading fail during Asterisk startup) and my phone (C7940) seems to be not supported in original chan_skinny :( PJ

Hi, the last 2 days i was working on getting the 7920 Phones to work with Skinny & Asterisk; however no luck (yet). Does anybody has a SEPDefault.CNF.xml and a SEP<mac>.CNF.xml handy for me ? it should be documented at the cisco page, but it isn't :-( I still have the issue that the 7920 spits out "No Service - IP Config failed" but Asterisk is giving me sign that the

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

Hello everyone! Anyone know if there is a way to use chan_skinny or chan_sccp to emulate a Cisco 7960 and talk to the CallManager. I realize it is intended to talk TO the phone, but I'm looking for an SCCP softphone solution (OS X). IPBlue has a client for Windows, yes. So, how much of a stretch can this be? Can't find any info on this anywhere. Tried Wiki, etc. Could someone please shed

I've got several SIP clients with dynamic IP addresses Asterisk has one public and one private IP address SIP clients might connect to Asterisk from either the internet or the private network (192.168.1.255) - they're portable By default, directmedia/canreinvite is enabled and Asterisk sets up direct media connections between clients. In this case clients on the internet can make calls

Hi, when using some non dynamic host eg. host=192.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or feature - I wanna disable the usage of directmedia for some peers with fixed ip but wanna allow it in general. Same with transport=tcp. [97]

Hi all, I've just built a new installation of CentOS release 5.3 (Final) and have installed both <http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz>Asterisk 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing complex - Pentium Dual core 2ghz - 1gb ram - 70gb

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to "directmedia=yes" but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am