similar to: Skinny error : Unable to create channel

Hi List! I'm trying to get the Kirk IP600 (DECT Wireless phones) to work with * using the Skinny protocol (chan_sccp doesn't work, the phones do not register and I don't know how to debug this). Basically the phones are able to place calls but not to receive calls. The extension is ringing for the calling party but the handsets do not ring. By putting the IP600 in debug mode and

I'm going to try, Thanks very much Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Remco Barende Inviato: luned? 30 gennaio 2006 20.04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Kirk IP600 Hi! Yes, it works (sort of) but I still have some issues.

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

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco

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

I have a Cisco 7970 that I am tying to get working. The phone powers up and registers but nothing else. I am using the development build. Any thoughts? The message I get when the phone powers on and registers. *CLI> skinny debug Skinny Debugging Enabled -- Starting Skinny session from 192.168.101.6 Recieved AlarmMessage Device SEP000F23FCA366 is attempting to register --

Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio

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

I have Asterisk up and running and it is working great with my SIP phones. However, I have some "Skinny"-protocol Cisco 7960s. Does Asterisk support the Skinny protocol? I've seen some references to Skinny in the software. If so, should I stick with Skinny with the 7960 or convert to SIP? If anyone has some Skinny confs they would send me I'd be much obliged. If I should

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

Hi list! My skinny phone can make outgoing calls but incoming calls just keep ringing for the calling end but the phone that actually should ring doesn't ring at all. I guess I have something messed up with the dial command. I have this in skinny.conf: [z4040] device=SEP000000000000 (actual number removed) nat=0 callerid="z4040" <105> callwaiting=0 mailbox=199 transfer=1

I have a bunch of 7910's that I managed to get registered with Asterisk 1.2.14: managed5*CLI> skinny show devices Name DeviceId IP TypeId R Model NL -------------------- ---------------- --------------- ------ - ------ -- test7 SEP0004C1878F8E 192.168.0.226 6 Y 7910 1 The problem is that the phone resets when I attempt to make a call from it or place a call to it. If I pick up I have

We unconditionally search for the EXTENT_ITEM_KEY for metadata during balance, and then check the key that we found to see if it is actually a METADATA_ITEM_KEY, but this doesn''t work right because METADATA is a higher key value, so if what we are looking for happens to be the first item in the leaf the search will dump us out at the previous leaf, and we won''t find our item. So

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind NAT that has one way audio. The called party cannot hear the calling party who's using the 7910. skinny.conf ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 0.0.0.0 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)

Hi i have to do the following setup: 1 - i receive a call on skinny protocol 2 - i forward the call to a sip user I think that the skinny phone must be registered on asterisk, in a particular extension, for example: [skinny_internat_ext] 987,1,Dial(SIP/user@host) And if the skinny phone dials 987 i make the call to SIP/user. But how can i do that if the skinny phone isn't registered to

I assume there are several people on this list that have Cisco Call Manager implementations under their belt.... We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base