similar to: Unable to create channel - CVS Broken?

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, I have seen a couple of scripts that should be able to remotely reboot the 79xx phones, but I haven't been able to make it work for my 7940. Anyone able to guide me in the right direction? I am running the SIP 7.1 firmware. -- 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

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

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 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

hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1!

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs & chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

Hello all, I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA) on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one B channel. But when I open a second B chan, the sound is choppy, with too long gaps, and the CPU load is too high (~50%). On the Asterisk's console I get these messages: -- Executing Dial("H323:4478",

Changes replicate to it, but not from it. vsc\VSC-DC02 DSA Options: 0x00000001 DSA object GUID: fe066b13-6f9e-4f3c-beb4-37df1292b8cb DSA invocationId: 8a2b1405-07b1-4d92-89dd-1d993e59e378 ==== INBOUND NEIGHBORS ==== DC=DomainDnsZones,DC=mediture,DC=dom vsc\DC01 via RPC DSA object GUID: da9bb168-47a0-4368-aff3-bf06d1b869d2 Last attempt @ Tue Mar 14

I believe the problem is a lack of outbound replication for non PDC emulator DCs. You'll notice isn't even trying because last successful was epoch (never) yet there are no errors. Inbound replication for this DC seems fine. [root at vsc-dc02 ~]# samba-tool drs showrepl [...]==== OUTBOUND NEIGHBORS ==== DC=DomainDnsZones,DC=mediture,DC=dom aws\AWS-DC01 via RPC DSA object GUID:

On 3/13/2017 2:15 PM, Arthur Ramsey via samba wrote: > Upgraded to 4.6.0 on all nodes. Still seeing the same issue. > > If I create an object on vsc-dc02, epo-dc01 or aws-dc01 DCs it doesn't > replicate. If I create it on vsc-dc01 (PDC emulator) then it does > replicate. > > On 03/13/2017 12:13 PM, Arthur Ramsey wrote: >> >> I believe the problem is a lack

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line