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Hi Barry Flanagan, Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19. Aug 11:06: > SIPP is probably what you seek. http://sipp.sourceforge.net/ > > Hope this helps. That looks pretty like what I'm looking for! Many thanks! Sincerely, Dominique Haeber

...ded in my example config. Well, it's working great now. Last question, I promise :P. Is it possible to change the date format? I want it in Norwegian. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22. februar 2006 13:52 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail Arne Morten Johansen wrote: > It's fixed now. Great! > > In "/etc/mail/ssmtp.conf", thi...

Yeah I did change those. I'm using 1.0.8 (Or was it 9?). It seams that the system overrides these settings? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 21. februar 2006 14:54 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail Arne Morten Johansen wrote: > Hi. I'm having trouble controlling the user info when sending e-mails > from asterisk...

...e drag a file from a Windows directory - say autoexec.bat and drop it onto a samba mapped Unix drive, when we then go to the Unix box and vi the file all the lines end in the DOS EOL character (^M) is there any option/configuration in samba to make it convert DOS text files to UNIX format? Brendan Flanagan Leyland Trucks Ltd, Croston Road, Leyland, PR26 6LZ Phone: 01772 625715 Email: BrendanFlanagan@Paccar.com

Hi all, i need to test how many calls can withstand an Asterisk server. Do you know any good tools to strain the server? At best, there are scripts that I can run on a Linux server. I thank you for your tips Sincerely Dominique Haeber

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

...r 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received iseqno 18 not within window 19->19 Any help much appreciated. -- -Barry Flanagan

...mLineOverride=YES") line was commented out. Removing that comment did the trick :) Now I only need to change the e-mail's title. Is that possible? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22. februar 2006 13:25 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail Arne Morten Johansen wrote: > As mentioned earlier I did try that. Someone suggested that there might be an issue wi...

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

Hello there. Just to let you know i have just put in production mode puppet on 41 freeBSD 4.7 virtual servers (yes not real servers but "jailed ones"). I had some little problems with facter at first but i have manualy upgraded the servers that where affected by the little "facter does not read his facts" problem i had. I should have missed something in my deploiement.

I've been using realtime for sip users information. I noticed that when you are doing this, if you do a 'reload' or restart asterisk, the information in a 'sip show peers' goes away. When I do this, MWI stops working. I always though MWI used the astdb file ('database show') to determine where to send MWI but it must be using 'sip show peers' because when this

...onf, and am using SetCallerPres(prohib) in my dialplan prior to the Dial command. No matter what I set SetCallerPres to the CID is still displayed. Is there something else I need to make this work? I can't just set the CallerIDNUM to null, as it is needed for billing purposes. Thanks. -Barry Flanagan

On Wed, 31 May 2017, Barry Flanagan wrote: > sngrep? Isn't sngrep a great tool? Since discovering it my use of tcpdump/wireshark has cratered. Being able to compare an INVITE that worked with one that didn't (with color highlighting) rocks. -- Thanks in advance, ------------------------------------------------------...

...Can I just change this without screwing up everything? Or is there a command to change user info or something? As you can see, I'm not so good in Linux. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 21. februar 2006 14:54 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail Arne Morten Johansen wrote: > Hi. I'm having trouble controlling the user info when sending e-mails > from asterisk...

Hi, I am able to reproduce the following panic on a number of Solaris 10 06/06 boxes (Sun Blade 150, V210 and T2000). The script to do this is: #!/bin/sh -x uname -a mkfile 100m /data zpool create tank /data zpool status cd /tank ls -al cp /etc/services . ls -al cd / rm /data zpool status # uncomment the following lines if you want to see the system think # it can still read and write to the

Dear readers, under xen 2.0.5, kernel 2.6.11.4-20a-xen (both suse 9.3) asterisk 1.0.9 could''t replay its sounds. Its sounds very brocken. Without xen it works fine. Would somebody help pls? best regards Stefan _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

...number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. -- -Barry Flanagan

Steve, would you be willing to share that "quick bash script"? James Cass <http://goog_987864563> jcass78 at gmail.com On Wed, Aug 19, 2015 at 12:11 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 19 Aug 2015, Dominique Haeber wrote: > > Hi Barry Flanagan, >> >> Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19. Aug 11:06: >> >>> SIPP is probably what you seek. http://sipp.sourceforge.net/ >>> >>> Hope this helps. >>> >> >> That looks pretty like what I'm looki...

Hi all, I''m trying to use acts_as_ferret and have run into a brick wall. My model is Page My controller is Pages_controller. When in console, I can search for contents, and find results. For example, when I search for "spam" it "finds" 7 results. => #<FerretMixin::Acts::ARFerret::SearchResults:0x2693934 @total_hits=7, @results=[]> you can do a