search for: hhandresen

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, Sometimes my asterisk server stops. (after a day or two) Last output from CLI is: -------------------------------- -- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120 -- Channel 0/26, span 1 got hangup -- Hungup 'Zap/26-1' voip1*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0).

Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from <IP-ADRRESS> When I google'ed this problem I can see other users also found this error (bug ?) But no-one

11-10-2004 there was a subject: Re: Where did USE_SIP_MYSQL_FRIENDS go?: on asterisk.user list. >All db specific code has been removed from the code in favor of the >currently-in-development "RealTime" method of configuration from >database. >You are most likely not using the 1.0 stable branch. >You need to use the new RealTime configuration method. And currently,

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

Hi, Yesterday I finaly got my handytone sip adaptor. It works.... But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to

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, 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, My asterisk fails and stops after running the reload command ~20 times (I'm testing) - is this a kown problem ? Therefor I wil reload only sip, extensions and iax, it works with sip and extensions, but it seem that there are no reload for iax - or what ? -- mvh. Hans-Henrik Andresen ------------------------------------------ Telefon for en flad 20'er - www.telefin.dk

Hi, When I make a call from my asterisk and it is passed thru another astrisk eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting as soon I get connected to the other asterisk, and not if the party on the other asterisk server pick up the phone. So IF the other party are not aswering the call at all, I still get Answerd and billsec in cdr. Whats wrong ? Can I do

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

Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel

Hi I'm running safe_asterisk, but get core-files in /tmp - how do I debug them ? I know gdb asterisk core.12370 and bt full But it didn't show anything usefull for me. Can anyone help me ? (Running asterisk 1.0.2 with ast_data /Hans-Henrik ----------------- Last from bt full: priority=200, callerid=0x81b8e90 "Dial", action=1134845864) at pbx.c:1384 e =

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

Hi After going from AST_DATA (RES_DATA) to realtime with mysql-driver my console is jam'ed with SIP Seeding peers from Astdb '000b8201XXXX' at 000b8201XXXX@81.146.XX.XX:35273 for 120 I got arround 4000 sipclients registered at that server and all the sip-client re-register every 120 sec. so the console is totaly fill'ed with SIP Seeding messages. Is it posible to not see

I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5

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, A'm about to set up a asterisk for 5000 users, and the customer had a 64bit server - can asterisk compile on that ? I will use a digium X100P for timing use will that do on a 64bit ? (I'm using SUSE91 kernel 2.6) What else ? Is it posible to have only one server for 5000 users ? I gues that it will be 5-700 sim. users only talking sip, and IAX2 to my PSTN-Gateway. The system is

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, 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