similar to: Unwanted conference with snom320 and asterisk 1.07 bristuffed

Under Advanced make sure this is set: Call join on Xfer (2 calls): to off ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tommaso Calosi Sent: Thursday, May 04, 2006 4:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unwanted conference with snom320 and asterisk

I have a preoblem with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2 firmware. All have same settings in the advanced panel. On 2 phones when I press the hold or transfer key nothing happens and * does not start the musiconhold. In the The hold and transfer keys are set as F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work ok. Any ideas?

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget

[general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to A realm=192.168.0.2 context = default ;Default for incoming calls [5549] disallow=all allow=ulaw allow=alaw allow=gsm type=friend ;(inbound and outbound calls accepted) secret=localphone ; obvious password for testing host=dynamic callerid=Jason White <5549> dtmfmode=auto mailbox=5549 ;(Asterisk VM-system's

I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the "disable call waiting" in asterisk as well, but I have not been able to find any

I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been

Hi all, My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls over ISDN2. Now I want to implement receiving incoming faxes into my setup so I did a google and some reading on the wiki. I got the spandsp 0.0.2pre10 package compiled and installed, patched asterisk's apps makefile and compiled * again. This all worked out fine. When integrating the RxFax into my dialplan the

Hi folks, I've already searched the mailing list but no one else seems to have my same problem. I'm using Asterisk with the following configuration: Fedora Core 4 (but I also tried Fedora 3) 1 Digium TE110P 1 TDM40B 1 HFC-S 'Cologne' bristuff 0.2.0-RC8o (zaptel 1.0.9.2) I compiled right, I can load kernel modules but when I try to unload wcte11xp module (the one

I hva ean HFC-S card in a box that I'm trying to get to work with bristuffed Asterisk 1.0.3. The box is an Athlon64 running a RHEL rebuild with a plain vanilla 2.6.10 kernel. I tried both APIC and NOAPIC mode. The installation went ok and does give output that seems correct SPAN 1: CCS/ AMI Buil-out: 399-533 feet (DSX-1) 2 channels and one D-channel Even though I've configured * to

Hi list! I'm using phones that emulate a Cisco 7940 with chan_sccp. When I was using Asterisk 1.0.5 (bristuffed) I never had any such message on the console. The phones do work. Is this a bug in chan_sccp or a feature of asterisk 1.0.6? Thx!

Hi, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob.

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Hi! I have 1.0.3-BRIstuffed and hfc-s card ztcfg says that card is configured (2 B and 1 D channel) but asterisk don't pickup any calls. Any ideas? Regards, Corvin

Hi, I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel : Linux laps1 2.6.13-15.7-smp #1 SMP Tue Nov 29 14:32:29 UTC 2005 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f . I get this : laps1:~/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc # make load make -C /usr/src/linux-2.6 SUBDIRS=/root/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc

I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there

I tried to get callerid working the normal way but the cid is never passed to the phone. It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in extensions.conf which I found in the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc Is this intended behaviour, or still a bug? It does work but it only shows one zero even though I have nationalprefix = 0

Hi, we've purchased new Beronet Octobri card and have problems loading modules for stock Asterisk for Debian Sarge (it has bristuff patches in it). when loading qozap it says that no multibri card was found although lspci shows it... There were quite some rumours about bristuff not liking other than junghanns cards, but don't know if something happened.... Anyone recently used this

Hi, I've upgraded my Asterisk to 1.0.7version patched with bristuff 0.2.0-RC8c. I'm using the same extensions.conf but it seems now include instruction doesn't want to work, here follows an extract: [inbound_menu] include => ins_exts exten => _X.,1,Answer exten => _X.,2,Wait(1) exten => _X.,3,Background(msg) exten => _X.,4,Background(3-sec-pause) exten =>

Hi list! I'm trying to get PrivacyManager working but for some reason it always thinks that CallerID is present (when it isn't). I get this on the console: == Primary D-Channel on span 1 up -- Accepting voice call from '' to '0711234567' on channel 0/2, span 1 -- Executing Ringing("Zap/2-1", "") in new stack -- Executing

>> Same problem with debian sarge on a dell and asterisk 1.0.7 from >> packages, unloading the module freezes the system, (rebooting the >> machine worked right), I installed zaptel 1.2beta and it seems to work, >> but I haven't really tested it, only loaded/unloaded/loaded and placed a >> couple of calls. >Interesting. The zaptel part of the bristuff