similar to: Skinny not working?

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

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)

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

Hi All, I'm trying to get a Cisco 7905g IP Phone to work with our Asterisk server but I'm having problems getting the phone to answer a call or make a call. I'm using the stable branch of the asterisk CVS on a RH9 box. I have got the phone to register with * and it retrieves it's extension number, date & time etc., but when I pick the handset up it's just dead silence

I admit that this might be some very basic question... How do I obtain Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3? Thanks, Adi

Hi all, I bought a couple phones for really cheap just for a simple solution. I'm trying to get a few 7910 to work with *. I'm just not sure how to get them to work. The 7910 just sits there "configuring IP" Here is a copy of my skinny.conf. the extensions.conf is default. I just want to bring the system up in default before a start making changes. Do I need to make

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

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, I've been looking at the various SIP VoIP service providers and their plans. I understand that Asterisk can be configured as a SIP client to access, for example, a BroadVoice account to access the PSTN and discount LD. I see that a lot of the features provided by SIP VoIP service providers are really not needed since Asterisk will provide them locally. I have no plans on dropping my

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

Hi, How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco claims both phones do SIP. I was strongly considering Polycom phones. However, it appears to be quite difficult to obtain support or firmware for Polycom phones. On the other hand, I find Cisco is very well supported. Thanks, Adi

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

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