similar to: Remote rebooting a Cisco 7940

Hi, I'm trying to compile zttool from the Zaptel lib, but I just can't find the sorcecode for Libnewt. Anyone got a link? Since i'm using LFS, I can't use precompiled packages. -- Med venlig hilsen / Best regards Michael L?jtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 S?borg Tel (+45) 3955 0700 - Fax (+45) 3955 0707

Hi, Just started to get this error after updating to the latest CVS. Asterisk dies if it can't create a channel - not so good. -- Executing SetCallerID("SIP/750-2550", "39660426") in new stack -- Executing Dial("SIP/750-2550", "CAPI/39660426:22179808") in new stack Jun 22 13:52:05 NOTICE[262161]: chan_capi.c:1172 capi_request: didn't find

Hi, I have a problem that I just can't figure out how to solve. I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

Hi, Just made a simple test to see how the two extensions (s and i) worked but for some reason I can't seem to make then act as I would like them to. I pick up the phone and dials 100 or 200 - and in the CLI it prints out what ever I have put in the Noop() If i dial any other number, nothing happens - no indication in the CLI. Souldn't the s or i context be activated when I dial a

Hi, I am trying to track down the reason to my problems with sending and reciving fax with my PRI and 2 TDM400P Cards: PSTN <-> PRI (E100P) <-> * <-> TDM400P <-> Fax Machine I have used Zapbarge to listen to the data stream, but I can't say if it really have some time slips - fax kinda noisy in itself. Using the zttool i saw the Sync source for the TDM are

Hi, I am trying to recice a fax with * using SpanDSP - but it doesn't create the output file. (See the bottom of log file). * Loads both app_rxfax.so and app_txfax.so fine. Also I can't make * autodetect an incomming fax call (yes I have enabled faxdetect=both in zapata.conf - though it's not a Zap device) Any ideas are welcome :-) Best Regards Michael L?jtnant System Details:

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Hi, I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver) to work with an asterisk box. The phone connects nicely to an external VoIP company (sipgate.de reportedly using asterisk themselves) but there is a strange problem with my asterisk: - Incoming calls via ISDN (chan_capi) to the Zyxel work perfectly. - Outgoing calls (capi/sip/iax) via asterisk are dialling out

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

(I posted this note on http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too) I tried to put together comments that were asked for on the P2000W. These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott for getting me started with it. DTMF keys work fine and are "read" correctly by Asterisk. It's important that you upgrade your phone to a modern version

P.S. I _had_ read the other posts that suggest changing to unicast sntp mode. This didn't help. I eventually setup a new ntp server on my LAN, and used it as a sync source. Everything seems OK now. Obviously a problem with my other ntp server. Cheers. -------- Original Message -------- Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon

This question is not actually * related, but please don't flame me! Is anyone out there using the 7.4 or 7.5 SIP firmware on their Cisco 79xx phones? I have a weird problem where my clock disappears after a period of time, and the only thing that will get it back is a reboot. Has anyone experienced this?

hi folks well, tinc is a really nice tool and we implemented it on 3 linux servers and 2 mobile clients (XP notebooks) so far. one of the 3 tinc servers is making troubles, when a connection is initiated from this server over a zyxel 642 adsl router out to the other 2 servers in the internet. the logfiles of the other 2 servers shows: > tinc[1398]: Received UDP packet from unknown

Hi All, My current 7940 phones use P0S3-06-3-00. I'd like to upgrade them so they're not massively out of date. I found a page at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx that gives some info, and using the cisco links there have tried to upgrade. According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades

I have asterisk 1.09 running with chan_sccp.20050701 on Fedora core 4 with kernel 2.6.12. I have a Cisco 7940 working with X-lite working on XP too, connecting to each other, Asterisk and FWD. The 7940 upgraded to SIP7.4 with xmlDefault.cnf.XML getting delivered via tftp using the loadInformation8 parameter method taken from voip-info.org ("Configuring Cisco 79xx phones with Asterisk"

If your phones support PoE, I have had huge success with Zyxel: http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00 5GRETMM/ref=sr_1_3?ie=UTF8&qid=1426296572&sr=8-3&keywords=zyxel+poe If you want to go even cheaper, I have successfully used these as well: http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN