similar to: 'Answered' at wrong time.

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

Hi, If I do this in extensions.conf exten => 411,1,Dial(IAX2/hhandresen@iaxtel/18005558355@iaxtel,40,rS(10)) the line is cut of in 10 sec., thats fine, but in CDR I got dst as T, and not 411. How can I handle this so I still get kicked of after 10 sec., but get 411 as dst in my cdr ? -- mvh. Hans-Henrik Andresen

Hi, I was thinking if it was possible to get this list as news ? It would be much easier that 'hotmail-account' /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of IAX2[2109@2109]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA

How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? /HHA

Hi I got a problem with asterisk 1.0.2 - it drops the calls, both sip<-->sip, and zap<-->sip. The conntions can stay for seconds to several minuttes, and then the connection just cut off. I can't see anything in the logfiles. (or dont know what to look at.) It drops several connections at a time, but not all. Where to start looking ?? /HHA

The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call is not answered billsec is set to duration, and calls are charged. I can't see any change I could have made to cause this problem. Is it something already known in Asterisk? /voipfc

Hello: I have an asterisk server answering SIP calls. Whenever a call comes, asterisk answers, plays a gsm file (information) and dials to another SIP phone. Using asterisk Master.csv file I only have one record, and don't know if the second call is answered. I only know this if: - The called phone is busy - The called phone doesn't answer in X seconds (the parameter in Dial) But I

Hi, How much bandwidth do I need for 1 conversation ? I know it depends on the codecs, in X-lite I can see a codec called gsm, and the grandstream aha analog/ip converter have a codec called 721. Doc. I have found the asterisk handbook, but only a draft from marts 2003 anything newer ? Guides/howtos are welcome as well. anyone have a php interface to accounting ? /HHA

We recently installed Asterisk 1.4.2 Tried to make calls using the Originate command (Asterisk Manager Interface) All of the calls have zero billsec in the CDR. Stumbled upon this: http://bugs.digium.com/view.php?id=8680 so I guess the fix is not yet in 1.4.2. Is this fixed in 1.4.3/1.4.4? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Hi, I got this setup. analog phone (ext7) ---> analog pbx ----- (ext 6 analog) voicemodem (ext 3 asterisk) ---- ttyS0/asterisk ---- sipphones q1: I got the voicemodem to work, but oneway only. I can talk from my analog phone, to my sipphone, but not the other way ? I know it only suppose to works in half duplex, but nothing come TO the phone. q2:

Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _________________________________________________________________ Find high-speed ‘net deals — comparison-shop your local providers here. https://broadband.msn.com

Hi, We are using SAMBA on Solaris and AIX and discovered a funny thing. (A bug ??) (We had tried on a Linux-box as well and have same result.) SAMBA AFTER VER. 2.0.0 returns the full path in both UPPER and LOWER case, eg. DRIVERLETTER:\DIR\DIR\DIR\filename.txt This is a problem for us do to Unix case-sensitive. Our devel.dep. wrote a litle program using this routine (From the helpfile from

I'm having an issue with cpu-spikes on the following setup Samba 3.0.24 as PDC with cups as the printing system on a Debian Etch 64 bit version Intel dual-core system My main problem is, that everything works just fine, there are no serious errors in the log files, even if i step the log level up to level 3. What happens is that the processes belonging to the guest-account have occasional

Regarding my previous post: "Playback audio before answered by a queue member" I added a ResetCDR() command at the middle: exten => XXXXXXXXXX,1,Background(audiofile) ;answers the channel immediately exten => XXXXXXXXXX,2,ResetCDR() ;clean slate exten => XXXXXXXXXX,3,Queue(Qname|tdn|||) ;new answer time written Looking at the CDR, the billsec is no longer the same as

We've got this configuration : Cisco as5400 --- asterisk main server ---- asterisk for cells ---- gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind

Hello Wine community, I have a problem with the HTML Help Work Shop compiler. It crashes with the newer wine version (0.9.54 up to 0.9.60) under kubuntu 7.10 For information, my Html work shop install is as said at this link: http://appdb.winehq.org/objectManager.php?sClass=version&iId=2978 1/ the command regsvr32 itircl.dll returns: Failed to load DLL itircl.dll. However, the dll is in

samba@samba.org Giulio Orsero <giulioo@pobox.com> Mike Quin <m.r.quin@stir.ac.uk> Matthew Halliday <matthewh@fesa.co.uk> Carsten =3D?iso-8859-1?Q?Nordstr=3DF8m?=3D Jensen <north@get2net.= "Carlos Vinueza M." <carlos@ecuadorexplorer.com> Eric Dahnke <edahnke@istreetlabs.com> Giulio Orsero <giulioo@pobox.com> Giulio Orsero