Re: [Asterisk-Users] TFTP server for GrandStream BT phones / need testing

Conrad Wood wrote: Does the patch add any functionality to atftp that tftpd-hpa[1] doesn't have? This patch adds only GS BT phones recognition funcionality. tftpd-hpa does not correct handle the OPTION parameters in the TFTP packet ;( At first I tired to implement it into tftpd-hpa, but after

[Asterisk-Users] TFTP server for GrandStream BT phones / need testing

for phones - custom ring tones for the phones You can find patch, source/unpatched/ and DEB for debian/sarge at http://projects.hudecof.net/linux/atftp/ best regards Peter Hudec -- Linux hackers are funny people: They count the time in patchlevels. -- Martin Josefsson

[Asterisk-Users] granstream, vlan, tftp

the configurtion and software images from TFTP. After the phone boots, all is working all rigt, becasue it used formaly saved configs from WWW. I tried two switched (cisco 2950, d-link des3526) best regards Peter Hudec -- It's so simple to be wise. Just think

Re: [Asterisk-Users] URGENT - micronet asterisk on h323

my debug is ./asterisk -vcg extension.conf is OK becaouce form ATA186 calls are working, aleso from cisco7905 and cisco as5300. And of course insde * it is working to. It doesn't matter, that the first is WAIT(1). It crashes any way. Peter Hudec Jeremy McNamara wrote: Peter Hudec

[Asterisk-Users] URGENT - micronet asterisk on h323

client to H323 network. please help me, this is urgent best regards Peter Hudec -- Company: PosTel a.s.Position: IP network manager Borska 6 Bratislava 841 04 mail: [EMAIL PROTECTED] www: [http://www.postel.sk] phone: [+421 02 50203169

[Asterisk-Users] which codec will be used ?

hello, my situations is as follows. In our comapny we are planing to have *. I'm testing it now. If we will buy G729 codec for * ... UA(SIP) - FW - (SIP)*(H323) - (H323)GATEKEEPER(H323) - (H323)AS5300 - world the following equipment speeks G729: *, GK ,AS5300. All call from UA to another

[Asterisk-Users] Call transfering, conferencing

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join

Re: [Asterisk-Users] Call transfering, conferencing

http://www.asterisk.org/index.php?menu=features - Call features - Call Transfer WipeOut wrote: Peter Hudec wrote: hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another

Re: [Asterisk-Users] Call transfering, conferencing

thanks, you didn't make me happy :( hudecof WipeOut wrote: Peter Hudec wrote: http://www.asterisk.org/index.php?menu=features - Call features - Call Transfer Yes, provided your phone supports transfer or you use the t or T options on your dial string and then use the # key to transfer

[Asterisk-Users] SIP behind NAT problem

Hello, my next problem is with SIP device behind NAT. First few seconds of the call are OK. Astrisk is sending the packets to the public IP address of the FW/NAT (62.152.224.3). But this change in 10 second and packets are send to the my public addres.(192.168.1.163). in the sip.conf for the

Re: [Asterisk-Users] BOTH UAs behind same FW/NAT

thanks for explanation. It does not solves this problem, but another one :) best regards hudecof Olle E. Johansson wrote: Philipp von Klitzing wrote: You will probably have to use canreinvite=no in the UA definitions in the SIP.conf for those two phones.. In your

[Asterisk-Users] BOTH UAs behind same FW/NAT

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is

Re: [Asterisk-Users] BOTH UAs behind same FW/NAT

WipeOut wrote: Peter Hudec wrote: hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15

[Asterisk-Users] Calls out of the PBX

hello, I have jsu configured my first Asterisk PBX and it works well. In our company we have alose one Cisco AS5300. How can I mmake Asterisk to forward calls, which have first digit 0 to that Cisco AS5300. Our gateway is allready configured to handla that calls. best regards

[Asterisk-Users] newbie - sip, pxb, ata, nat

. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for testing end point to the public IP. best regards Peter Hudec -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] phone: [+421 2 50203163] icq: [99518783] gpg: [http