similar to: Resolving digit strings using pound/hash.

Hi all, I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32 machine, and can't find any problems. So, I decided to upgrade my home pbx. All went well until I tried using my S101 to talk to the IVR. Some times, the first one or two digits get through, but eventually a digit will get stuck, playing continuously until the call is terminated. I have confirmed this

As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. In the ChangeLog for 1.4.18 a bug (11735) was mentioned. I do seem to remember that in 1.2, it wasn't possible to send a message to ones-self. This bug fix apparently corrects that situation. Well, I guess it would, if only it

Hi, When I build the same asterisk package that I build on i386 on x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same set of installed packages. What should I be looking for in the output of ./configure to get a clue of what might be missing? TIA. -- Bill in Denver

Hi everyone, I have a problem on samba running on RHEL 3 kernel 2.4.21 with RHEL 3 samba packages - samba-3.0.7-1.3E.1. The problem is, that when people open excel files (with office 97 version of excel) the file modification time is changed (even if the file is only opened and excel closed without saving). This same problem does not occur on the [homes] share (in every test untill now it never

Hi, I am translating the voice files of voicemail now. I don't know if the POUND and # are the same key in the telephone's keypad. If they are same, how could we understand the following message: %vm-msginstruct.gsm%To hear the next message press 6, to repeat this message press 5, to hear the previous message press 4, to delete or undelete this message press seven, to quite voicemail

Hey, is http://www.dovecot.org/patches/1.0/quota.tar.gz supposed to work with maildir quota's for virtual mailboxes? Using it with dovecot-lda (cvs) results in DSN Your message was automatically rejected by Sieve, a mail filtering language. The following reason was given: Internal quota calculation error Thanks, Lieven -------------- next part -------------- A non-text attachment was

Dear all, I've used trixbox to compose a custom IVR, and I've defined the input of pound key(#) leading to the repetition of the parent announcement. But each time the pound key is pressed, file "dir-intro.gsm" will always be played. Can any one tell me the reason? By the way,if I want to modify the build-in IVR, which configuration file should be edited? Thanks in advance.

Hi, Can anybody confirm the following problem: Sequence: bash-2.05# grep ankerpos /usr/samba.moonrock/private/smbpasswd ankerpos:921:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX:[U ]:LCT-0 bash-2.05# /usr/samba.moonrock/bin/smbpasswd -c /usr/samba.moonrock/lib/smb.conf ankerpos New SMB password: Retype new SMB password: bash-2.05# grep ankerpos

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

Hey, I'm doing a setup of dovecot (cvs) + postfix + dovecot lda (cvs). Is there any solution to the /var/run/dovecot/auth-master access restrictions, prohibiting the user running dovecot-deliver from communicating with the master (aside from the dirty and unsafe hack to reset the permissions of the /var/run/dovecot dir after starting dovecot)? Thanks, Lieven

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

On Mon, 18 Jan 2016, Ethy H. Brito wrote: >> how to flush user input before READ()? How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hello, I''d like to try out HTB for traffic shaping. I have a test machine with Redhat 7.2 (kernel 2.4.10), I''ve downloaded kernel 2.4.17, applied the patch from http://luxik.cdi.cz/~devik/qos/htb/, recompiled the kernel (checked the option to use HTB in make menuconfig), and rebooted with the new kernel. When I try to add some rules with HTB (f.i. tc qdisc add dev eth0 root

On Mon, 18 Jan 2016 16:09:17 -0200 "Ethy H. Brito" <ethy.brito at inexo.com.br> wrote: > On Mon, 18 Jan 2016 09:38:52 -0800 (PST) > Steve Edwards <asterisk.org at sedwards.com> wrote: > > > On Mon, 18 Jan 2016, Ethy H. Brito wrote: > > > > >> how to flush user input before READ()? > > > > How about a read() to a dummy variable

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal