search for: minkov

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip config [phone1010] type=friend secret=damian auth=md5 nat=no host=dynamic reinvite=no canreinvite=no qualify=1000 dtmf=...

...number_wvm,1004,SIP/1004' Here is what i have in extensions table : id context exten priority app appdata _____________________________________________ 1 sip-internal 1004 1 Macro dialnumber_wvm,1004,SIP/1004 -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: damian@space-comm.com http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, &q...

...om 'damian <sip:damian@10.1.1.11>' failed for '10.1.1.11' Asterisk and Sip phones are all in one network , no nat. Here is the Config in sip.conf [phone1010] type=friend host=10.1.1.11 auth=md5 nat=no reinvite=no canreinvite=no qualify=1000 dtmf=inband callerid="Damian Minkov" <1010> username=damian

...ad: RTP: Received packet with bad UDP checksum venera*CLI> /usr/sbin/safe_asterisk: line 83: 26579 Segmentation fault asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 139 Asterisk exited on signal 11. -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: damian@space-comm.com http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, &q...

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,

I have * box on machine with external ip address and internal one I'm tring to register to it from to machines - one from innternet ( everything is ok - in sip.conf nat=yes)\ and the other one is in the internal network (in sip.conf - nat=no ) and it say 403 Forbidden? Any Ideas ? here are the logs and configs From the external SIP Client whic registers.

No Caller ID comes from the FXO line ( The caller id is on and is working with a standard phone) in zapata.conf everything looks fine usecallerid=yes hidecallerid=no When the call comes in there are some warnings in Asterisk Console -- Starting simple switch on 'Zap/4-1' Jun 22 11:20:24 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 17 Jun 22 11:20:24 NOTICE[213006]:

How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

hi all, Is the echo cancellation in the last svn version working ? As I'm running the testecho but with no change in the output ! Maybe my recordings are not correct. how ca I get correct input files or can some one provide such test files. Thanks in advance damencho

I have Wildcard TDM400P - so after compile i'm loading zapatel and wcfxs - kernel modules. My question is does I need other module in order to work with the FXO and FXS. ( beacause in other document i've noticed that there was writen to load and wcfxo - module ) but when i try this I get an error. voipgw:~# modprobe wcfxo /lib/modules/2.4.26/misc/wcfxo.o: init_module: No such

I have TDM400P , with 1 FXS and 1FXO I'm tring to forward all incoming calls to a SIP phone in the context where all calls from the fxo come i have : exten => s,1,dial(SIP/phone1000,5) the phone rings but when i answer the sip phone ( phone1000 ) is connected but the phone from which i'm ringing still rings. Here is the log from asterisk : *CLI> -- Starting simple switch on

I have problem setting the language for the SIP channel - Sip.conf I have set there language=xxx and I've recorded some mesages for voice mail in xxx language. But when they are played they are in en. But this works for language=xxx in zapata.conf (When these messages are played to the FXS) Any suggestions ? I've tried to put SetLanguage in extension.conf But it causes me some

We have ISDN PCI Adapter from Billion(using mISDN). When we connect it to the PSTN the incoming calls are OK. When we try to make a call we can't do that because the ISDN port is Down: CLI> misdn show port 1 BEGIN STACK_LIST: * Stack Addr: 40400001 Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Idx: 0

Hi all, does anybody has ran echocancellation successfully on windows ? I have a problem playing audio there. If you play a wav file no mater how I play it after comparing the captured data with the original file there is always difference in the audio length (for 18 sec of audio there is more than 250 ms of difference). Which of course breaks echocancellation. I've tried playing the audio

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk