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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

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

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

On 30 April 2017 at 16:54, Tech Support <asterisk at voipbusiness.us> wrote: > I thought this was a non-commercial list. > > Yeah, I wouldn't mind so much if it had actually answered the original poster's query. "Switch to our proprietary solution and we can offer you this proprietary solution" isn't a contribution, it's an ad. -Barry > >

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

...;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 looking for! Many thanks! >> > > Anoth...

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

Hello, I have a Dell PowerEdge 2850 that periodically (6 times in the last 24 hours) freezes, requiring a power cycle in order to come back. The machine is running a Xen 2.0 stable source install. Here is the appropriate Grub entry: title Xen 2.0 / XenLinux 2.6 kernel /boot/xen-2.0.gz dom0_mem=131072 module /boot/vmlinuz-2.6-xen0 root=/dev/sda1 ro console=tty0 Dom0 boots happily, and new

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

Hi, Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk. Recently I have been getting errors from Asterisk due to corrupted From: headers, which appear to be caused by uac_replace. Here is a section of the debug log: Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DBG:uac::restore_from_reply: removing <From: <sip:lenc_domain.com@sip.domain.com>;tag=635c3ce6 > Mar

Hi, Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on certain extensions. I have usecallingpres=yes in zapata.conf, 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

Hi I have an installation based on Daniel-Constantin Mierla's excellent Kamailio 3.3 / Asterisk 10 Realtime document ( http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) but have come across an issue which is a potential problem. In this installation all SIP clients register with Kamailio, and the registrations are forwarded to Asterisk. This means that all

Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new 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. --