Is anyone aware of any way of changing the contact header on a call? We are sending 911 calls to a provider and they require that the contact be the call back number. I tried:Set(PJSIP_HEADER(update,contact)=)But the came back with:No headers had b..
Author : Dovid Bender
Home » Articles posted by Dovid Bender (paged 4)
Sorry in advance that I am emailing the users list and not the biz list Ithink I will find my target audience here. We are looking to hire a consultant to help us figure out an issue. We are having what seems arerandom load issues with bare metal bo..
All,I am stuck with a specific install using chan_sip and Asterisk 11.25.3. We have nat=no which from what I understand means that Asterisk will go by whatever it sees in the SDP and not look at the source IP+port from where the traffic is coming fr..
All,I built a system which allows people to call a phone number and listen to various online media streams (train yards, radio stations etc). I use ffmpeg + MusicOnHold to play the streams. The system also allows callers to hear pre recorded content.Norma..
We have a client that is looking for a CPCI or ATR system. If someone from sales could get back to me that would be greatly appreciated. I can be reached at 848-210-0001 or 914-600-2000.Thanks in advance.Reg..
Does anyone know of any projects that would allow you to use Redis in place of AstDB? By in place of I dont mean for what Asterisk needs but to store values. For instance for CNAM currently we need to use an AGI to connect to redis to pull CNAM. So..
We have a box up and we are starting to see a lot of Exceptionally long queue length queuing in the logs. From all the research so far it seems like this leads to their systems crashing and being unreachable. In our case the box remains up and ta..
All,I vaguelyremember someone at Astricon making the case for having multiple containers/vps each running asterisk vs using asterisk direct on bare metal. Something about getting better performance. Does anyone have any insight on this?TIA and stay safeDo..
1) Is there any reason why max_pseudo_channels defaults to 512? I want to increase it by default but at the same time dont want to outsmart the developers if they had a good reason for it.2) I had a look at http://lists.digium.com/pipermail/asterisk-users/2014-March/282607.h..
Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non admins f..