similar to: Can't talk on Cisco VIP 30 using Chan Skinny

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is having problems with Voicemail. They can listen to their voicemail but on the weekend it stopped delivering messages via email. The only thing I can notice is that the permissions for the files on teh voicemail directories are created with no permissions at all. Here is the listing on one of the mailboxes: 4 -------rw- 1

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

I have an issue getting any incomming calls When the phone rings something picks up and gives it a fast busy. There is no one using Zap/2 it does the same thing with voicemail and voicemail 2 you can see the console output below, I would love any help anyone could shead on this issue, Michael NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2 (Ring/Answered)... --

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

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).

Hi everyone, I've set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I've already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? One interesting artifact is that at

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! 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

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 there, I know this was discussed many times already, but after trying to get my 30VIP to work for a week, I think I might be doing something wrong. I'm trying to set it up with chan_skinny. Haven't tried chan_sccp yet, but since there are many of you already doing it with skinny, it should be working also in my case. I read and used all the hints and samples I was able to find on 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)

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

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