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 debuging the code I give it up. The tftpd-hpa reads the parameters from the TFTP packet as they are in the packet and after each option it do the associated command. The packet from GS BT looks like --- cut --- Opcode: Read Request (1) Source File: boot.bin Type: octet Option: blksize = 1024 Option: tsize = 0 Option: timeout = 4 Option: grandstream_MODEL = BT-100 Option: grandstream_NAT = 1 Option: grandstream_ID = 000b8203e0e9 Option: grandstream_REV_BOOT = 001.000.001.000 Option: grandstream_REV_PHONE = 001.000.006.007 Option: grandstream_REV_VOC = 001.000.001.000 Option: grandstream_REV_HTML = 001.000.000.049 Option: grandstream_REV_RING1 = 001.000.000.000 Option: grandstream_REV_RING2 = 001.000.000.000 Option: grandstream_REV_RING3 = 000.000.000.000 --- cut --- So it reads ... Opcoce - do something Source File - send the file to the client ... Reads the Options from packet. So in the time or sending requested file, there I have no information about the phone. I was too lazy to correct this ;( hudecof > > [1] http://packages.debian.org/stable/net/tftpd-hpa > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- "Linux hackers are funny people: They count the time in patchlevels." -- Martin Josefsson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TFTP server for GrandStream BT phones / need testing
hi, I known, that this is not * related, but a lot of members of this ML uses GS BT phones. I have patched the atftp serber to recognize the TFTP OPTION, whis these phone send during boot. Patch includes - another locations for configs, firmware and ring tones - different FW versions 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] granstream, vlan, tftp
hi all,
I known taht this ML is about *, but lot of you are using BT telepnones.
I'm using FW 1.0.6.7 for all phone. This firmware support VLAN tagging for
QoS on the Layer2. I use it to separe the PHONE network form the PC network,
which are in PC connector in the BT.
And I also use the TFTP server for provisioning (software, configuration).
The problem is, if you set the VLAN for QOS. When the phone is booting, it's
send
untagged pakets to the switch and therefor this paket is placed to the
defualt vlan (configured on the switch) and wrong network (PC). So the phone
does
not retrieve 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
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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 wrote:
--- CUT ---
-- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new
stack
== Spawn extension (postel, 169, 1) exited non-zero on
'H323/ip$62.152.225.18:52434/20702'
--- CUT ---
First off you are going to have to provide more debug than just that and
secondly a Wait,1 doesn't do anything but wait one second. I suggest
checking your extensions.conf file, you have a missing priority number
or whole exten line.
Jeremy McNamara
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[Asterisk-Users] URGENT - micronet & asterisk on h323
hello,
my situation is
h323gw - gatekeeper - asterisk - SIP client
my problem is, that I can't make call from h323gw, when this GW is
Micronet (sp5004). A
--- CUT ---
-- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack
== Spawn extension (postel, 169, 1) exited non-zero on
'H323/ip$62.152.225.18:52434/20702'
--- CUT ---
On the other side, when the h232gw is Cisco ATA186, Cisco 7905 or Cisco
AS5300 all is working good.
I'm using standart h323 modul, which is included in the *.
OH323 modul allways crashes.
I can make call from SIP client to H323 network.
please help me, this is urgent
best regards
Peter Hudec
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[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 endpoing go through *, because of mixed SIP and H323 sihnalization. If the UA does not known G729 (known GSM, G711), which codec wil be used between * and AS5300 ? Will * translate GSM codec (or other) to the G729. If yes, how to get it work? best regards hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 phone(X-Lite) is [998] type=friend username=998 secret= canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=998 nat=1 callerid=0650199802 Can anybody explain me, why the dest IP will change from public one to the private one ? best regards hudecof - CUT - DEBUG[540687]: File rtp.c, Line 388 (ast_rtp_read): RTP NAT: Using address 62.152.224.3:8000 DEBUG[540687]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 4160, ms is 540 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 DEBUG[540687]: File rtp.c, Line 388 (ast_rtp_read): RTP NAT: Using address 62.152.224.3:8000 DEBUG[540687]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 6576, ms is 842 DEBUG[540687]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 7848, ms is 1001 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 - CUT - -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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.. CLI> show application dial ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 another person to our talk . I haven't found this in any manual :( hudecof You won't find in in any Asterisk manual becasue these are not features of Asterisk, they are features on the phone.. The phone needs to support transfer and if you want conferencing without using "meetme" then you need a phone that supports conferencing.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 another person to our talk . I haven't found this in any manual :( hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 case you want the opposite: canreinvite=yes A try to sort out these kind of opposite messages: When asterisk connects two SIP phones, it tries to be in the middle of the media path, to have the RTP stream go through Asterisk. This way, Asterisk may send early media and error messages over audio. When the call is connected, asterisk can send SIP re-invites and change the path of the media stream, so that media flows directly between the two phones instead of going through Asterisk. This is canreinvite=yes In your situation, for calling between the phones, you propably don't want the media stream to go SIP UAC -> NAT -> Asterisk -> NAT -> SIP UAS (canreinvite=no) Instead SIP UAC -> SIP UAS (canreinvite=yes) However, I'm unsure if you can have a canreinvite=yes, since you may want asterisk to be in the media path when calling outbound... Also note that some devices does not support SIP re-invites (according to the Asterisk handbook) I'm a bit on thin ice here, so if I'm wrong - please, list, correct me so we can sort this out. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards hudecof You will probably have to use "canreinvite=no" in the UA definitions in the SIP.conf for those two phones.. I have this so Also make sure you have enough badwidth between the UA's and the Asterisk server to sustain 2 calls.. 100Mbit/s full duplex Later.. Still the same ;( hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie - sip, pxb, ata, nat
hi all,
I don't know how to setup asterix to work as PBX.
If I want just basic configuration with 2 SIP phones (snom, ata), what
all I have to write in the configuration files, or respectively in the
configuration of ata and snom ?
If there is any good documention available, send me URL too.
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
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