similar to: Fax, T38 and NAT

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

Hallo, ABCTI (an open-source CTI client for Asterisk) has moved to beta stage. Find it on: http://abcti.sourceforge.net For the first time, we now have windows installers that actually work ;-) We would appreciate any feedback you can give. Regards, -- o

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

I am attempting to use rsync to copy a large filesystem from an HP-UX server to a Linux server with more than enough filespace. This operation fails. A small directory from the same HP-UX server can be transfered just as expected. The HP-UX server is the source. It has 1Gb RAM - the output of bdf for the volume the source files is on is: Filesystem kbytes used avail %used Mounted

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys

Hi all, I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine via a T.38 enabled trunk.? I've got t38pt_udptl = yes faxdetect=no in my sip.conf file.? The ATA has all of the T.38 options turned on, echo cancellation is off, as well as silence suppression off.? The only configured codec is u711.? When the user tries to send a fax, it gets to the point where it

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi! Did play around with queues and need some help. I thought that MEMBERINTERFACE and MEMBERNAME should be set to the ?device? the call was queued to not the device that called the queue, or do i miss something? Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on 2011-01-31 13:38:23 UTC 0317998985 calls Kinna (0320209030) Tomas Ekman (SIP/0317998972) receives the call but

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hi! I try to get rid of some part of CALLERID(name) but I cant realy figure out a way to do it. For example: CALLERID(name) = "Martela (fax)" I am just looking for the part before ? (? in my case ?Martela?. I can?t serch for ? ?, could be many ? ?, but only one ? (?, thought i could do something like: exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1}) But that gave me

Hi! Have a weired problem with Avaya 9650 phones: extensions.conf exten => 0317998975,hint,SIP/0317998975 exten => 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1) exten => 0317998975,2,Hangup() exten => 0317998975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 0317998975-INUSE,2,Hangup() exten => 0317998975-NOANSWER,1,VoiceMail(0317998975 at

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

Hi I'm debugging why echo cancellation does not work in my app using the DUMP_ECHO_CANCEL_DATA flag. What format does the data I receive have, looks like 16bit integers just dumped but when plotting it in MATLAB I do not get a raw audio data signal I expected. Kind Regards //Gunnar Gunnar Karlsson HotSwap Stockholm AB Landsv?gen 39| SE-172 63 Sundbyberg, Sweden Mobile +46 739

Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But "core show applications" doesnt show me any "fax applications" and when I try to receive a fax:

Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.

Hi all! First of all i would like to say that i am not a Samba4 guru so my question may be "stupid". I have 2 Samba4 servers up and runnning: Server 1: netbios name = PDC workgroup = GBG realm = GBG.INPUTINTERIOR.SE server role = domain controller Server 2: netbios name = PDC workgroup = MLM realm = MLM.INPUTINTERIOR.SE server role = domain controller Here comes my

Hi All ! Recently in the mailing list there was the question about benchmarking LLVM. I was told that LLVM get's benchmarked in the nightly test. While this is true, I wanted to have a tool to compare LLVM against other compilers, so I wrote a little python program (attached) that filled my need. It is completely outside of the LLVM makefile framework, but this stems from the fact that I

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP