[Asterisk-Users] Asterisk Compilation with H323 working on it
Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Compilation with H323 working on it
Hi Did it work well with Netmeeting from Microsoft ?? Thanks for answer. Carlos. On Thu, 20 Oct 2005 14:41:38 +0200, Bohuslav Coufal wrote: I did use it on Debian and now use it on FC4 and H323 is working good on both systems. Im using asterisk own h323 driver. Bob. From: [EMAIL PROTECTED] [mailto:asterisk-[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 2:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Compilation with H323 working on it Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos.Carlos Arnt Key soluções em Internet Av. das americas 500 bl 03 sala 204 Tel: (021) 2492-1666 Voip rede mundial: 9000 ou 9500 E-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routes IPSEc And Asterisk.
Hi Folks, Really need a help here. I have 3 network places Let´s say : A , B and C Using Freeswan (Ipsec) i make the point A see and interact with point B (Now both networks see each other) Everything is perfect, but i have in point B now a C Network that comes over Router. Point B com see and interact with Point C , but point A can´t In number : Point A = 192.168.2.0/24 Point B = 192.168.1.0/24 Point C = 192.168.3.0/24 Over ipsec i can ping from A to B and from B to A. From B i can ping C and from C ping B. But from A i can´t see C . Because C is not a VPN over Ipsec, it is only connected in my B network with address (192.168.1.254) I insert a route : route add -net 192.168.3.0/24 gw 192.168.1.254 Then everyone from point B can see Point C. How make point A see C too ?? I need this because i send asterisk packet over ipsec . Thanks alot for helping out !! Carlos. Ps. Point C don´t have IPSEC it´s just a router connected to point B in swith over lan. Thanks. Again. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial 0 to outbound
Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0 to call for example a pstn the user will ear the line sound to dial out. I read lot's of doc's but can't find nothing explaining this method. Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canreinvite=???
Hi, everyone ! Looking at this explanation : When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other. So if i really understand this using this option i can make the RTP packets flow from one device to another when they connect leaving only the SIP to asterisk . So for example if then I put my Grandstream with a real ip address and use * with a real ip address i can make my calls from nufone flow direct to my grandstream leaving my * bandwidth free . Like this : Grandstream begin call SIP--- Asterisk | | - Nufone. Open RTP Channel Grandstream Real IP -- Nufone IP Right ?? If i'm right , i try this and with tcpdump see the even with everyone using real ip's, the RTP still going over asterisk using my bandwidth . (Note, I force grandstream to use the same codec then Nufone, G729) Can someone give-me some light ?? ;) Can i make this ??? Use asterisk only to begin the call and let the RTP flow over the client and nufone network ?? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP Packets going over caller and calle !!
Hi, I Have a problem here, if anyone know a method to avoid please tell me . Using * with the option canreinvite=yes i can in theory tell to my * box, send RTP Packet directly from one Sip device to another one, then "In Theory", i will not use my own internet connection. So this mean that will a lower connection something like "512/512kbps", i can have lot's of channels connected and using my * just to bridge then . Thats correct ?? But if all people is under a NAT ?? Like 2 sip devices using * box connect over my * box but that two is under NAT ?? Will work anyway ? All RTP Packets is flow using their connection ? Not mine ? If anyone understand this nonsense question please help me ;) Thanks alot !! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best VOIP PSTN Provider.
Hi People, I wondering here, who is the best VOIP PSTN Provider to use with my * box ? That has good prices, good quality (Use ex. G729 codecs) etc ? I want to make calls to Europe, Asia etc with cheap prices and good quality in the sound ;) Did anyone has a good tip for me ? For use with IAX2 or SIP and with g.729 codec. Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto: Installing Asterisk and ISDN on Fedora Core 1
ISDN Carlos Arnt [EMAIL PROTECTED] Diretor de Informática. Divisão de Tecnologia e Desenvolvimento TI. Intellissence do Brasil. http://www.intellissence.com/brasil Tel:(+55)-(21)-(3908-4667) Tel(Direto):(+55)-(21)-(3905-1561) Cel:(+55)-(21)-(9169-8537) --VoIP Contact Method-- World VoIP Pin/Code: 31872 Uk/Japan VoIP Pin/Code: 17009881356 - "Thinking is the hardest work there is, which is the probable reason so few engage in it". - Henry Ford - On Fri, 25 Jun 2004 11:46:28 +0100, Asterisk wrote: I've managed to install and run Asterisk on a Fedora Core 1 server using the fritz avm ISDN card, and thought I'd share how it was done. This worked for me: Server: Dell 6450 Quad Xeon 700 2Mb Cache 4GB Ram 2x18GB SCSI (Mirrored) ISDN card (fritz!pci). This is packaged as a BT ISDN card. * Install FC1 from CD. Select only server components, development environment and kernel development * Yum update Important - you must do this - it installs a new kernel, not the buggy release kernel. * Reboot * Install atrpms-55-1.rhfc1.at.i386.rpm (from http://atrpms.net/dist/fc1/atrpms) * Install kernel-module-fcpci-2.4.22-1.2194.nptlsmp-03.11.02- 3.rhfc1.at.i686.rpm (from http://atrpms.net/dist/fc1/fcpci) * vi /etc/capi.conf should contain a single line fcpci - - - - - - * modprobe fcpci * lsmod Should now show fcpci and kernelcapi loaded * capiinit * Get Asterisk from a shell, cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login (anoncvs is the password) cvs checkout zaptel libpri asterisk * Build Asterisk cd /usr/src/zaptel make clean;make install;make config cd /usr/src/libpri make clean;make install;make config cd /usr/src/asterisk make clean;make install;make samples * Get chan_capi (from http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.4a.tar.gz) * Extract chan_capi cd /usr/src tar -zxvf /tmp/chan_capi* (assuming the file was downloaded into /tmp) * Build chan_capi cd chan_capi make clean; make install; make config * Add chan_capi into Asterisk by changing modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so load = chan_modem.so load = res_musiconhold.so following two lines added load = res_parking.so load = chan_capi.so noload = chan_alsa.so [global] chan_modem.so=yes following line added chan_capi.so=yes * Start Asterisk asterisk -c (should be no errors) * Place a call to the ISDN line. As I said, this worked for me. The real problem I had before was getting the drivers for the isdn card to work. However, downloading them from the atrpms site worked first time no errors! Please feel free to tear this apart if you want. Julian. ___ Asterisk-Users mailing list[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and ISDN
Hi, I'm trying to use Asterisk with one ISDN TELES 16.03 c PnP Card (ISA) Now i can call Asterisk with the Modem i4l driver etc But need more information to make a better config and also know how call this card to make outgoing calls and receive incoming calls. I know how use with a Zap card, but in ISDN mode i can't make work well.. If anyone know a better place to learn how config this well will be a good help. Thanks alot. Ps. Sorry any english problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ebay X101P Card. CRAP!!!
Hi People, I know that this is a Digium forum, and actually i will buy cards now from Digium too. But a have just a question. For test purposes and of course save some money a buy from Ebay a " Mercury M/N: AMI-IA92 card." With this card Asterisk work well - my linux appear like "Tiger Jet card". But i notice some problems: 1 - Dropped calls (I try avoid with Busycount=x) Works but sometimes drops anyway. 2 - Freeze the line. (Sometimes this cards freeze the line and just become normal after restart asterisk). 3 - Voice problems, I notice that this card has some problem with Sound because sound become unstable for the person that listen in the other side. (Appears using a VAD making the call very bad) So anyone has the same problem ? Did anyone know a fix for that ? I will buy a Digium "Official" card just to test and see the diferences. Anyway more info from everyone will be very good. Thanks alot. Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel problems with Silence Detection
Hi people, There is any way to control silence detection in Zaptel ?? I have a x101p card, and sometimes the sounds dont come. I notice that this is a Silence Detection in the card how can i avoid this ? Another things, g.729a has silence detection ? Or Asterisk do this ??? Using a sip i can have any option to this ? And finally , Where i find a document that show me all options for use in sip.conf, iax.conf , zapata.conf etc ? Thanks alot ! Carlos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two FXO Cards answering at different times.
Hi all, Anyone know how put my X101P cards to answer at different ring times ? Like x101P(a) Answer at 3 rings x101p(b) Answer at 4 rings My * it's connected into a PBX thats when receive a call send to two lines at same times a ring. (So i must have a way to just put one channel to answer not both at same time) The behavior is that when i answer one the other channel answer too. Then my client receive 2 calls , one its the normal call the other just a signal of busy. Did anyone know how stop this problem ?? (Don't say buy a new PBX ok!!) ;) Thanks alot . Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP CBQ BandLimit HELP!!
That's great.
Maybe i will ask a nonsense question.
Let go then :
Sip uses RTP right ? So open a SIP channel 5060 i have alot of RTP packets .
Did i don't need mark this RTP packets too ??
I mean IAX2 use RTP ? In you script i see that i MARK IAX2 then i can control the rate and give
to my VOIP connection more priority then the other services, (Web,Mail,FTP etc), always making the voip connection the best possible.
That's great, awesome!
But in your script ? If i change this :
iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 4569 -
j MARK --set-mark 0x1
iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 4569 -
j RETURN
to this ?
iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 5060 -
j MARK --set-mark 0x1
iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 5060 -
j RETURN
All my SIP connections will receive now the best priority in the link ??
Sorry the question again but and the RTP packets ???
Can with this script give from my link 512up and 512down 90% to Voip when used and let the rest (Web,FTP,Mail), with the rest 10%, so when the VoIP finish give more to the others services???
Anyway thanks alot for the answer !
Carlos.
On Tue, 1 Jun 2004 09:30:29 -0400, Andrew Kohlsmith wrote: On Tuesday 01 June 2004 05:44, joachim wrote: Do you have a working firewall ruleset for HTB, optimized for voip ? Here, for your viewing pleasure, is my htb script. I am *positive* it can be improved upon. I found I had to put the bulk traffic in a separate HTB "branch" or otherwise it would tend to borrow from the VOIP branch way too early and cause a lot of stutter. Again, this isn't to be taken as an official, proven and perfect system. I am welcome to any suggestions or improvements. As you can see from the diagram I am planning on doing some more fine tuning, but it works pretty damned well as is. My next post will be the Cisco 2610's configuration on the other side of the link. Regards, Andrew #!/bin/sh TCOP="add" IPTOP="-A" if [ "$1" == "stop" ]; then echo "Stopping..." TCOP="del" IPTOP="-D" fi # +-+ # | root 1: | # +-+ #| # ++ # | class 1:1 | # ++ # | | | # ++ ++ ++ # |1:10| |1:20| |1:30| # ++ ++ ++ #| # +++ # ||| # +-+ +-+ +-+ # |1:100| |1:101| |1:102| # +-+ +-+ +-+ # 1:10 is the class for VOIP traffic, pfifo qdisc # 1:20 is for bulk traffic (htb, leaves use sfq) # 1:30 is the class that interactive and TCP SYN/ACK traffic (sfq qdisc) # 1:20 is further split up into different kinds of bulk traffic: web, mail and # everything else. 1:100-102 fight amongst themselves for their slice of excess # bandwidth, and in turn 1:10,20 and 30 then fight for any excess above their # minimum rates. # which interface to throw all this on (DSL) IF=eth2 # ciel is 75% of max rate (768kbps) # rate is 65% of max rate # we don't let it go to 100% because we don't want the DSL modems to have a ton # of packets in their buffers. *we* want to do the buffering. RATE=576 CEIL=640 #RATE=450 #CEIL=500 tc qdisc ${TCOP} dev ${IF} root handle 1: htb default 102 tc class ${TCOP} dev ${IF} parent 1: classid 1:1 htb rate ${RATE}kbit ceil ${CEIL}kbit tc class ${TCOP} dev ${IF} parent 1:1 classid 1:10 htb rate 64kbit ceil ${RATE}kbit prio 1 tc class ${TCOP} dev ${IF} parent 1:1 classid 1:20 htb rate 64kbit ceil ${RATE}kbit prio 2 tc class ${TCOP} dev ${IF} parent 1:20 classid 1:100 htb rate ${RATE}kbit tc class ${TCOP} dev ${IF} parent 1:20 classid 1:101 htb rate ${RATE}kbit tc class ${TCOP} dev ${IF} parent 1:20 classid 1:102 htb rate ${RATE}kbit tc qdisc ${TCOP} dev ${IF} parent 1:10 handle 10: pfifo tc qdisc ${TCOP} dev ${IF} parent 1:100 handle 100: sfq perturb 10 tc qdisc ${TCOP} dev ${IF} parent 1:101 handle 101: sfq perturb 10 tc qdisc ${TCOP} dev ${IF} parent 1:102 handle 102: sfq perturb 10 tc filter ${TCOP} dev ${IF} parent 1:0 protocol ip prio 1 handle 1 fw classid 1:10 tc filter ${TCOP} dev ${IF} parent 1:0 protocol ip prio 4 handle 4 fw classid 1:100 # IAX2 prio 0. iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 4569 - j MARK --set-mark 0x1 iptables -t mangle ${IPTOP} PREROUTING -p udp -m udp --dport 4569 - j RETURN # everything else goes into lowest priority (best effort). iptables -t mangle ${IPTOP} PREROUTING -j MARK --set-mark 0x4 iptables -t mangle ${IPTOP} OUTPUT -j MARK --set-mark 0x4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] VOIP CBQ BandLimit HELP!!
Hi all, Reading about CBQ on internet i can say "I dont understand well" ;) So anyone that has a good background can help me out with this simple question ? I just want priorize my UDP packets to always has 90% of my link when use a VOIP connection with asterisk. My asterisk run in the same machine then my firewall. How then can i : 1 - Mark the packets with iptables then i will know TCP and UDP packets then come in and out 2 - Use CBQ to put a prio=1 in the UDP Packets then i will always know that when a VOIP conn start will always have the best rate of my link. I think i know how mark the packets with the Iptables. iptables -t mangle -A PREROUTING -p tcp -j MARK --set-mark 9000 iptables -t mangle -A PREROUTING -p udp -j MARK --set-mark 9002 and iptables -t mangle -A OUTPUT -p tcp -j MARK --set-mark 9001 iptables -t mangle -A OUTPUT -p udp -j MARK --set-mark 9003 I think that i mark all UDP and TCP packets. So i just need use a CBQ RUle (Now it's the worst) Honestly i dont know .. So let's see. DEVICE=eth0,10Mbit,1Mbit RATE=112Kbit WEIGHT=1Kbit MARK=9000 etc etc I use an 256kbits(Down) - 128Kbits(Up) ADSL connection Then i have PPP0 and my eth1 for my internet net. Just need put the best priority to all UDP Packets forcing the rest of services like SMTP/POP3./HTTP etc that use TCP in the low priority Can anyone help me ? Because i think my Voip has a poor quality because this (Heavy use of mail and http services). Thanks alot for helping out. Carlos I ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Hangup always..
Hi all, Did anyone see this problem. I have two X101P in my asterisk Box. When i receive a call, everything goes Ok, but after a short period of time i receive a hangup .. Sometimes happen at 1:50min others at 05:00min but always happend. Did anyone see this before ?? How can i make the cards use a different call pickup ring ? Like Zap1 pickup a call in 2 rings Zap 2 pickup a call in 4 rings Thanks for helping out ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grab phone call ?
Hi Let's say i have a call to a extension 115. But i'm under the extension 118 how take the call from 115 to my extension using * ?? Thanks alot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk prepaid debug
What kind of prepaid agi did you use ?Could you send me the page ? How install or where find ? Thanks alot. On Mon, 19 Apr 2004 20:07:14 -0600, Julio wrote: My Asterisk prepaid debug is: - Hungup 'Zap/2-1' Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler -- Playing 'prepaid-enter-card-num' (language 'en') Urgent handler -- Playing 'prepaid-you-have' (language 'en') Urgent handler -- Playing 'digits/4' (language 'en') Urgent handler -- Playing 'digits/hundred' (language 'en') Urgent handler -- Playing 'prepaid-dollars' (language 'en') Urgent handler -- Playing 'prepaid-enter-dest' (language 'en') Urgent handler -- Playing 'prepaid-dest-blocked' (language 'en') Urgent handler -- Playing 'prepaid-dest-unreachable' (language 'en') Why 'prepaid-dest-unreachable' ?? Thks. Regards - Original Message - From: Martin Christian Koch To: [EMAIL PROTECTED] Sent: Monday, April 19, 2004 4:05 PM Subject: [Asterisk-Users] spandsp/rxfax terminates asterisk Initial handshake sounds fine, but asterisks dies before receive of the fax. Here is the log : Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 30 36 37 37 36 31 36 35 20 35 34 2b 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: "+45 56167760" DCS: 83 00 46 20 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1700.00 (64) Training error 29.095569 Training succeeded (constellation mismatch 25.504344) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 1 Start rx page asterisk in realloc(): warning: junk pointer, too high to make sense Oh dear! CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.04 (64) Training error 26.487284 Training succeeded (constellation mismatch 27.123313) Fast carrier trained Segmentation fault (core dumped) Anyone ? Thanks, Martin Min mail er beskyttet af SPAMfighter 3174 spam mails er blokeret indtil videre. Hent gratis SPAMfighter i dag!
[Asterisk-Users] HT 286 Any information about will be great !!!
Hi, Did HT-286 Bypass calls from normal PBX and Asterisk PBX to analog phones ? To be more precisely, can i receive both call from this two kind of tecnologies using HT-286 in my office ? I dont want change my OLD PBX (That works great) with Asterisk and lose investiment etc. So i think in use HT-286 and then use both at same time to receive public calls and Internet call over my desk with the same phone. Can i receive ? It will works ? If i'm into a Internet call and the old pbx send a call , did the HT-286 send a busy signal? In the other way too? Did HT-286 talk GSM ? Well anyone that have it, tested IT and enjoy have it ! Please answer. Thanks alot. Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best VOIP Analog adapter ???
Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog phone . Anyone know some adapter that make this miracle ? Thanks alot, Carlos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000 SPA-2000, Help Question.
Hi, Can anyone tell me if SPA-2000 sipura, talk GSM and bypass from a normal PBX and Asterisk to a analog phone ? I want to use SPA-2000 Adapter in my office with both my * and the old PBX that has in it. Can sippura byppass the calls from both to my analog phone ? So i can receive calls from the pbx and * and make call from my analog to both too , the old pbx and * .. Thanks alot Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream and HT-286
Hi, Did GrandStream Voip-phone and HT-286 Analog adapter talk using GSM 6.01 Codec ?? In tests using asterisk this codec it's the best for my kind of connection, did both Hardwares use this codec to talk ? In page they don't mention this. Thanks alot. Ps. ILBC ?? Talk too ?? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About HT-286
Hi, Did HT-286 a good answer to put into my company and with both * and my old pbx use VOip and Normal telephones without change any kind of structure ? Like with HT-286 can i just plug it using the RJ-45 part in the * and the RJ11 in my old pbx and have both working with my normal telephone ? And make this . My normal old pbx has channel from 20 to 80 (Normal local phones etc) with * i can then put (90 to 100) and has the VOIP channels both with the same phone right ? Can anyone have the answer for this question ? Thanks alot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P CallOut Problems !!
Hi all, I just now receive the FXO X101P Card but can't at any way make then call out. I can hear the signal, even call but always receive from my local operator error that or the number don't exist or need more numbers. I play alot with txgain and rxgain, but none help me out. Being honest i try alot 5 hours and none !!! I'm using asterisk in his sample configs. I mean i call out using 1234 etc.. Zapata.conf is Ok Zaptel.conf is ok (I follow the Digium faqs, then for a good person that show-me this in the Asterisk IRC) ( Using here is an Asterisk 7.1) Did anyone know a txgain and rxgain from Brazilian lines ? (I'm trying with Vesper operator) Did i need make something more ( i know that need) :) Please could someone with lot's of time help-me out here with this simple question ? I just wanna call out too !!! Thanks alot ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P CallOut Big Problem.
Hi all, I just now receive the FXO X101P Card but can't at any way make then call out. I can hear the signal, even call but always receive from my local operator error that or the number don't exist or need more numbers. I play alot with txgain and rxgain, but none help me out. Being honest i try alot 5 hours and none !!! I'm using asterisk in his sample configs. I mean i call out using 1234 etc.. Zapata.conf is Ok Zaptel.conf is ok (I follow the Digium faqs, then for a good person that show-me this in the Asterisk IRC) ( Using here is an Asterisk 7.1) Did anyone know a txgain and rxgain from Brazilian lines ? (I'm trying with Vesper operator) Did i need make something more ( i know that need) :) Please could someone with lot's of time help-me out here with this simple question ? I just wanna call out too !!! Thanks alot ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
IntelliFAX Como funciona. Sua rede fica ligada a internet bem como seus fornecedores. Você passa o fax normalmente que se encotra ligado em nosso PABX Virtual e seus fornecedores via Internet usando uma versão light do sistema poderão receber de qualquer canto do mundo este fax em seus faxes. Basicamente e um servidor que fica ligado a internet e a seu Fax. Podemos vender pra aquela agencia de turismo ! Abraços vamos conversar fui. []'s On Wed, 14 Jan 2004 09:13:25 +0800, Steve Underwood wrote: Jason Penton wrote: Hi All I have just a quick question regarding app_txfax for Asterisk. When I send a fax from asterisk to a traditional fax machine connected to asterisk via the digium analog card everything works perfectly. However the same fax machine on the public telephoine network results in errors (looks like some sort of training error). My asterisk box is connected to the pstn using an ISDN card. I don't mind trying to fix this myself but I am puzzled by the different behavior experienced when the fax machine is on the digium card and when it is connected to our public carrier, and therefore have no idea where to start. Would someone (Steve Underwood ;-) )mind at least putting me on the right track so I can address this issue? Thanks in advance Steve Jason I don't know why this would fail. An ISDN card should be properly synchrinised to the PSTN, and uses A-law or u-law. That should be enough to get a good path from the txfax program to the FAX machine. Do you have some kind of codec mismatch in your system? Can you try attaching an analogue phone in parallel with the FAX machine, and listen to the audio? You are probably familiar with how a FAX machine normally sounds, so you can probably recognise the bad distortion a codec issue would cause. The other thing to listen for is clicks. If there is a timing problem, and you have even a single sample slip in the audio stream, a modem will not work. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
Só que existe um problema ! O sistema deles * não funciona com o seu !! O seu e desenvolvido em Windows (Delphi) enquanto que o deles e Linux !!! O * e totalmente VOIP o seu no momento fala um protocolo louco !! Me liga então. Ps. Ao invês de usar sua conta e o forum quer por favor, usar e-mail direto ! Abraços. On Wed, 14 Jan 2004 09:13:25 +0800, Steve Underwood wrote: Jason Penton wrote: Hi All I have just a quick question regarding app_txfax for Asterisk. When I send a fax from asterisk to a traditional fax machine connected to asterisk via the digium analog card everything works perfectly. However the same fax machine on the public telephoine network results in errors (looks like some sort of training error). My asterisk box is connected to the pstn using an ISDN card. I don't mind trying to fix this myself but I am puzzled by the different behavior experienced when the fax machine is on the digium card and when it is connected to our public carrier, and therefore have no idea where to start. Would someone (Steve Underwood ;-) )mind at least putting me on the right track so I can address this issue? Thanks in advance Steve Jason I don't know why this would fail. An ISDN card should be properly synchrinised to the PSTN, and uses A-law or u-law. That should be enough to get a good path from the txfax program to the FAX machine. Do you have some kind of codec mismatch in your system? Can you try attaching an analogue phone in parallel with the FAX machine, and listen to the audio? You are probably familiar with how a FAX machine normally sounds, so you can probably recognise the bad distortion a codec issue would cause. The other thing to listen for is clicks. If there is a timing problem, and you have even a single sample slip in the audio stream, a modem will not work. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpeeX and IAX2 ??? Work or not ??
All progs that i see only use GSM codec format to IAX2. What program can i use to test Speex Codec after all ? What i need put into iax.conf to force just use this codec ? "dissallow = all" " Allow = SpeeX " ??? I try IaxComm thinking that he uses Speex, but only call GSM too. Xlite can't use SpeeX i try too. Please could someone help me ? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpeeX and IAX2 ??? Work or not ??
Yep I know that. I just say X-lite because even in Sip can't work. But I see alot of people saying that use IAX2 with SpeeX codec well. I just be one of then too :) On Mon, 29 Dec 2003 08:34:32 -0500, Andrew Thompson wrote: - Original Message - From: "Carlos Arnt" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 8:25 AM Subject: [Asterisk-Users] SpeeX and IAX2 ??? Work or not ?? Xlite can't use SpeeX i try too. Xlite is a SIP client, not IAX(2). - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download
I put like the readme.txt say the code in my web page , put the OCX in the same Directory, but not work. Did has any problem ?? Thanks The code: html head meta http-equiv="Content-Type" content="text/html; charset=windows-1252" titleNew Page 1/title /head body OBJECT ID="diax" CLASSID="" CODEBASE="" /OBJECT /body /html On Sun, 21 Dec 2003 23:46:21 +0200, Dan wrote: Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any browser. It works only with IAX2, so you can test it with IAXTEL too (supported, just needed to dial the number). Please send me your feedback and features request for further development of both standalone and ActiveX versions of DIAX. Best regards, Dan P.S. Unfortunatelly it works only on IBM PC compatible computers (not Mac's or Pocket PCs) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best SIP PHones to buy ?
Hi People, Can anyone help-me here with a simple question. I wanna buy a Sip Phone, but what is the best and cheap one ? I see alot of messages about, grandstream , snow etc etc. So for use with my * system, what sip phone is the best ?? Can him be used behind a nat system etc ? Or with a Broadband connection etc ? Thanks alot for help!! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSN 4.7 and Asterisk.
Hi All, I'm just testing * with MSN 4.7 and works great, but when i try to call using the dialpad of MSN all my number into * appers twice. Like 100 when i try appears 1 .. Like a echo in the number. I'm using rfc2833 because inband just crash ... sip.conf. [msn] type=friend host=dynamic username=msn dtmfmode=rfc2833 callerid=0001 accountcode=msn nat=no canreinvite=no qualify=200 Someone can help-me ? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSN MESSENGER 4.7 with Asterisk -SOMEONE HELP HERE PLEASE!-
Hi all, I just trying to test MSN 4.7 that has SIP. Because with him i can use a video and voice transmission and * . But when i try to call someone using the DIALPAD of MSN, when i insert any digit into * the numbers appears twice !! like this. channel 456 appears in asterisk 445566 How can i fix this ? Please its very important !! Thanks alot for any help !!! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and voice recog support ?
Hi, Calling the FWD, i see a feature a little different. I don't call any number, but TALK with the system and they go to others parts of the showed menu. There are any way to make the same with * ? Where are the link that i'm talking about. http://fwd.pulver.com/callme.php?userid=5 Thanks Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Group dial codes ?(Newbie question)
Hi All, Using asterisk and extension.conf can i make a group dial code ? Like this. Ie. Let's say i have a group called directors. Only People in this group can dial to a external number like 800. How can i make this possible in asterisk ? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
Dan, I think i understant what he's talking about. In netmeeting if you put into your command line ie. START - RUN. Something like callto://192.168.x.x you can call another netmeeting. So you can put this into a webpage with lot's of peoples callto's and call then just clicking into a link. It's a registry stuff, I think he's asking you to put the same option into DIAX, so people can use: callto: tel: or maybe : diax://192.168.x.x and open the diax dialing directly the specific ip or address. If i'm wrong in my tip sorry for that :) Just trying to help. Carlos Arnt []'sOn Tue, 11 Nov 2003 09:30:41 +0200, Dan wrote: Hi, - Original Message - From: "Masakazu Nakano" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 11, 2003 2:18 AM Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download On Mon, 10 Nov 2003 17:40:02 +0200 "Dan" [EMAIL PROTECTED] wrote: and need 'callto:' support :-) Why you need this? Give me an example. 1) netmeeting and cuseeme are already support. 2) make a call easier with groupware and/or portal. ( 'tel:' is good too ) I really do not understand what do you mean... Can someone else explain what is about? What is 'callto:' support? What is 'tel:'? What about groupware? Thanks, Dan ___ Asterisk-Users mailing list[EMAIL PROTECTED] http://listsdigium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
My explanation it's ok. Dan, it's great what he think in use. So into a groupware app he can put into ex. the name of the person a link to call then over diax too. But to be honest your ActiveX version will be more helpfull than just using a link. hmm, but it's a good idea too . Carlos []'sOn Tue, 11 Nov 2003 16:58:31 +0900, Masakazu Nakano wrote: On Tue, 11 Nov 2003 09:30:41 +0200 "Dan" [EMAIL PROTECTED] wrote: Hi, - Original Message - From: "Masakazu Nakano" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 11, 2003 2:18 AM Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download On Mon, 10 Nov 2003 17:40:02 +0200 "Dan" [EMAIL PROTECTED] wrote: and need 'callto:' support :-) Why you need this? Give me an example. 1) netmeeting and cuseeme are already support. 2) make a call easier with groupware and/or portal. ( 'tel:' is good too ) I really do not understand what do you mean... Can someone else explain what is about? What is 'callto:' support? What is 'tel:'? What about groupware? oh... I'm sorry. Now I wish to making SOHO platform with twiggi and asterisk. twiggi is very cool LAMP + IMAP based groupware. http://www.twiggi.org/screenshots.php 'callto' is like that for M$ Windoze. http://msdn.microsoft.com/library/default.asp?url="">- US/netmeet/nm3_1l4o.asp and tel: is implementing some mobile cellurer carriers (such as i- mode). using 'make a call'. in Germany. http://sten- schmidt.net/imode/taglist.html mack ___ Asterisk-Users mailing list[EMAIL PROTECTED] http://listsdigium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
Well that's easy then . Look here : Windows Registry Editor Version 5.00 [HKEY_CLASSES_ROOT\callto] @="URL: DiaxTo Protocol" "EditFlags"=hex:02,00,00,00 "URL Protocol"="" [HKEY_CLASSES_ROOT\Diaxto\DefaultIcon] @="\"C:\\Program Files\\Diax\\diax.exe\",1" [HKEY_CLASSES_ROOT\diaxto\shell] [HKEY_CLASSES_ROOT\diaxto\shell\open] [HKEY_CLASSES_ROOT\diaxto\shell\open\command] @="rundll32.exe msconf.dll,diaxToProtocolHandler %l" Then now you can call Diaxto://address or name etc. And not use the callto command. Carlos. On Tue, 11 Nov 2003 15:44:38 +0200, Dan wrote: Hi, - Original Message - From: "Peer Oliver schmidt" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 11, 2003 3:31 PM Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download Dan, Hi Carlos, I ain't Carlos, but here are my thoughts :) Sorry for my mistake...;-) I think that the two can be reladed, in order to use Internet Explorer for DIAX (as ActiveX) when you enter to run something like: diax://192.168.x.x, even from a browser. Let's now thing at the syntax of such a command. You must call through an * server, so the syntax must be somethink like: diax://user:[EMAIL PROTECTED]/extension I would think, all you want is the dial string. I would not include the authorization as that is already being stored in DIAX. Also, the URI should not be diax:, as there already are URI definitions for this kind of szenario, like callto: for example. So, I would envision something like this: callto:[EMAIL PROTECTED] or callto:1234 basically, the same syntax used within DIAX to call someone. That way, an intranet webpage could contain a link to a customer callto:001-714-668-8877 or maybe to a colleague callto:[EMAIL PROTECTED] -- The only problem is that callto: is assigned to NetMeeting by default... ,... and I don't want to change that. BR, Dan ___ Asterisk-Users mailing list[EMAIL PROTECTED] http://listsdigium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
Yep, that's right. On Tue, 11 Nov 2003 15:06:19 +0200, Dan wrote: Hi Carlos, - Original Message - From: Carlos Arnt To: [EMAIL PROTECTED] Sent: Tuesday, November 11, 2003 2:36 PM Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download So into a groupware app he can put into ex. the name of the person a link to call then over diax too. But to be honest your ActiveX version will be more helpfull than just using a link. hmm, but it's a good idea too . I think that the two can be reladed, in order to use Internet Explorer for DIAX (as ActiveX) when you enter to run something like: diax://192.168.x.x, even from a browser. Let's now thing at the syntax of such a command. You must call through an * server, so the syntax must be somethink like: diax://user:[EMAIL PROTECTED]/extension I'm right? Thanks, Dan ___ Asterisk-Users mailing list[EMAIL PROTECTED] http://listsdigium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (ast_rtp_read): Unknown RTP codec 72 received
Hi, Please someone know what can be this message ? "(ast_rtp_read): Unknown RTP codec 72 received" Always i try to use the X-lite, this message appears. What is the 72 number ? what means ??? I use sometime and never see this, now from nothing just appears.. Thanks for helping. Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xphone Beta 1.01 ?
A simple question. I found an old Xten Xphone Beta 1.01 that has Sip capabilities. So i try with my asterisk, but he always try to login using this format: sip:[EMAIL PROTECTED]:5060 And say that registration failed. Someone see this kind of thing before ? I try using Xten Lite and everything goes Ok. I just try to see whats happend, because this same program is used in pocket pc's i think .. Can someone help help ? This is the log : File chan_sip.c, Line 5203 (handle_request): Registration from 'sip sip:[EMAIL PROTECTED]' failed for 'ipaddress' (I change the real name, and ip here in this mail for *name* and *Ipaddress* of course) :) Thanks for helping out! Carlos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk over VPN.
Hi People, Let's take a look in this diagram : Part A - Server running VPN IP ie.192.168.10.1 Part B - Client running over the VPN with internal IP ie. 192.168.10.2 -- From network A i can reach B. Use all programs - Share Printers , aplications, using Netmeeting etc.. Then i make this in the same server of the VPN i put Asterisk PBX. (Network A) Running SIP in the same network (the network below the server, all machines can login etc perfeclty and talk with each others). In the Network B . All machines can't connect to Asterisk ... Just if i point all to the External Address of the VPN Server that has asterisk... In the log i can only see the registrations using the external address of the client (VPN) not the Internal one . Ie. Using 200.300.200.100 not 192.168.10.2 Well i'm using X-lite to talk and works great . I make the same test using two machines one server with Asterisk and the other just a Windows CLient. If i point the Windows client to take the external address he login very well. If i try over the Internal address i can ... My question is , in the VPN Rules all TCP and UDP ports are open. I can even share printers and files etc in both machines, why i cant then use Asterisk to talk with my computer in this case inside the VPN ??? Thanks alot for helping . Carlos . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Dan, It's a great program, Just a question, it's open source right ? Can i see the code ? I'm a c++ programmer too with some time to spend now (On vacation) :) So can i help ? Did you plan put in the page a source code for people download too ? About the ACTIVEX idea it's great too !! Again nice job !! Carlos. On Thu, 6 Nov 2003 09:31:11 +0200, Dan wrote: HI, When I first load the Gui, I get to see directory displayed on the right hand (bottom) below RX/X/TX etc info. You mean first 12 memories.. Once a number is dialied, this place is used for Volume control but it would be nice to see the same directory all the time so that dialing would be easier. In order to keep the window as small as possible and still have the full functionality, the form is changed depend on current status. When you're in a call, you can not dial another number, so this is the reason that the memories are not available. Did you think about keeping the Dial and Hangup buttons as permanent feature instead of switching back and forth? Sometimes, I also see a delete button which may not be necessary? You must take a closer look at the functionality. The two function buttons depend (again) on the cuirrent status. They can display DIAL/DELETE, HANGUP/-, REDIAL/-, REJECT/- When you start enter a number using keypad, after the first digit you et the DELETE button too, in order to be able to correct the numbet. Each click on this button deletes the last digit. When the number is dialed from the memory, DELETE button delete the whole number. I cannot imagine something simpler than that... Anyway, that is my feedback so far. i will try using some more functions and let you know if I find something strange. Please check it closer. I still work on the detailed help file which will be available during the weekend. By the way, I had liked the idea of multi-line phone as a drop- down action as suggested by someone earlier. Default could be a single line with basic features. In this moment, the phone is very close in functionality with a full featured single line analog phone (with callerid, callwaiting, callwating callerid, and so on) I do not see an imediate reason to have more than one line for a standar phone. Do you? Keep it up. I'll do it... Thanks for your feedback and best regards, Dan ___ Asterisk-Users mailing list[EMAIL PROTECTED] http://listsdigium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF x-lite
Yep, that's right in RedHat 7.3 too ! Now works great! Thanks alot. On Fri, 31 Oct 2003 19:12:11 +0300, Shoval Tom wrote: I fixed it. finally. I was missing suidperl. Apparently it isn't installed on redhat 9.0 You can download it from www.redhat.com and install it. And voile everything works. From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Friday, October 31, 2003 5:45 AM To: [EMAIL PROTECTED]; asterisk-[EMAIL PROTECTED] Subject: RE: [Asterisk-Users] DTMF x-lite Well now it's two with this problem. I try everything but don't work until now.. try chmod +s try chmod a+s etc etc etc ... Try 4755 etc. None works.. Put the /var/spool/asterisk/vm with apache.apache AND nothing.. So if anyone has a tip, please help. Another question is, did asterisk have any kind of authentication level ? Thanks alot ! Carlos. On Fri, 31 Oct 2003 02:22:34 +0300, Shoval Tom wrote: I've managed to gather that the cgi problem as appears in the httpd error_log is that it can't do setuid. I've searched the web for the last couple of hours and tried almost everything I could find, and I still can't get suexec to work. Can anyone help, please? I know this probably is a newbie question, but the voicemail web interface is a great selling point for the ones upstairs. Thanks a lot for any answer. Shoval From: [EMAIL PROTECTED] [mailto:asterisk- users- [EMAIL PROTECTED] On Behalf Of Shoval Tom Sent: Friday, October 31, 2003 12:00 AM To: asterisk-[EMAIL PROTECTED] Subject: RE: [Asterisk-Users] DTMF x-lite Well, found the answer for the DTMF problem, and guys, the voicemail is G R E A T !!! The answer was use rcf2833 for dtmfmode, not inband as suggested earlier If someone can help me resolve the cgi problem, I'd be forever indebted From: [EMAIL PROTECTED] [mailto:asterisk- users- [EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Thursday, October 30, 2003 11:29 PM To: asterisk-[EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF x-lite Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says "Premature end of script headers: vmail.cgi" Shoval ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://listsdigium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF x-lite
Well now it's two with this problem. I try everything but don't work until now.. try chmod +s try chmod a+s etc etc etc ... Try 4755 etc. None works.. Put the /var/spool/asterisk/vm with apache.apache AND nothing.. So if anyone has a tip, please help. Another question is, did asterisk have any kind of authentication level ? Thanks alot ! Carlos.On Fri, 31 Oct 2003 02:22:34 +0300, Shoval Tom wrote: I've managed to gather that the cgi problem as appears in the httpd error_log is that it can't do setuid. I've searched the web for the last couple of hours and tried almost everything I could find, and I still can't get suexec to work. Can anyone help, please? I know this probably is a newbie question, but the voicemail web interface is a great selling point for the ones upstairs. Thanks a lot for any answer. Shoval From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Shoval Tom Sent: Friday, October 31, 2003 12:00 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] DTMF x-lite Well, found the answer for the DTMF problem, and guys, the voicemail is G R E A T !!! The answer was use rcf2833 for dtmfmode, not inband as suggested earlier If someone can help me resolve the cgi problem, I'd be forever indebted From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Thursday, October 30, 2003 11:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF x-lite Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says "Premature end of script headers: vmail.cgi" Shoval ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VMAIL.cgi
Hi All, I'm using now * a few months, perfectly with sip phones (soft and hard), but does anyone can explain why my VMAIL.cgi when used don't show any mail in the mailboxes ? I can login and use it but even with .wav's or .gsm in the proper directories he dont show nothing. All times (0 Inbox) etc etc Thanks Alot ! Carlos Arnt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
