similar to: SIP registration

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename.

Hi Masakazu, I have reproduced the Speex high frequency noise issue on a Blackfin 537 by sending 'silence' into the Speex fixed point encoder. By recording the Speex decoder output and plotting the spectrum using Audacity the following two frequencies. 1596Hz at -31.3dB 3200Hz at -48dB These two frequencies are present in both Narrow Band and Wide Band modes. I've tried a number

Originally posted by: mailto: Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I

I am running Asterisk CVS-04/16/03-18:57:13, and mpg123-0.59r It all sounds great and it plays at the correct pitch and speed. However at the end of the file it simply does nothing. It does not go on the the next step in the extension.conf nor does it hang up. It just sits there. During play I have two processes running for the mp3 stream: root 6300 6299 8 22:32 ?

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

Hi Jean-Marc, Thanks for the quick response! I am resending this mail hopefully in plain text. The following three version have been tried and all of them have the high frequency tones. speex-svn-14525 speex-1.2rc speex-svn-15540 I've isolated this down to the Speex encode/decode by sending silence (i.e. all zeros) to the encoder and displaying the decoder output in addition to

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

Hi Isamar maybe I think disclose your code to CVS is best and fast :-) mack > > Hi folks, > > I'm trying now to play with fsk_modem.c and callerid.c > to get the Japanese callerid working and I already got to make some > steps.. > I don't know if anybody accomplished that already... but > Since two or more minds think better than one, send private messages >

Hi all. I found a connection error from sipphone.com. It seems 'realm based authentication' by sipphone.com. any ideas? Regards. mack

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

** THE RETURN ADDRESSES ON THIS LETTER HAVE BEEN SET TO PREVENT MAIL ** ** LOOPS IN THE EVENT YOU ARE RUNNING SOFTWARE WHICH AUTO-REPLIES TO ** ** INBOUND MAIL. WORLDCOM WILL NOT SEE ANY REPLY SENT TO THIS LETTER. ** Dear Customer, We have received your message ("[vorbis] VorbisExt 0.3.0pre") and forwarded it to the appropriate personnel for processing. To help us provide you with

** THE RETURN ADDRESSES ON THIS LETTER HAVE BEEN SET TO PREVENT MAIL ** ** LOOPS IN THE EVENT YOU ARE RUNNING SOFTWARE WHICH AUTO-REPLIES TO ** ** INBOUND MAIL. WORLDCOM WILL NOT SEE ANY REPLY SENT TO THIS LETTER. ** Dear Customer, We have received your message ("[vorbis] VorbisExt 0.3.1 released") and forwarded it to the appropriate personnel for processing. To help us provide

Hello, It appears there is a limit in the number of criteria that can be put into the Aggregate sum function. (It looks like it is 32). My code is; HSfirst=aggregate(count,

Hi, i've setup samba on one of our solaris boxes, it was working fine, but suddenly now it does not start. here is the error message from the log.smb file: [2000/02/11 09:42:51, 0] locking/shmem_sysv.c:sysv_shm_open(566) Can't create or use semaphore [1]. Error was No space left on device [2000/02/11 09:42:51, 0] locking/locking.c:locking_init(174) ERROR: Failed to initialise share

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

Release Announcements --------------------- This is the latest stable release of the Samba 4.17 release series. Changes since 4.17.7 -------------------- o? Jeremy Allison <jra at samba.org> ?? * BUG 15302: log flood: smbd_calculate_access_mask_fsp: Access denied: ???? message level should be lower. ?? * BUG 15306: Floating point exception (FPE) via cli_pull_send at ????