In my experience this is usually caused by REINVITES.Disable reinvites (aka directmedia in recent Asterisks) and see if that helps. > —–Original Message—– > From: email@example.com > [mailto:firstname.lastname@example.org..
all, Im trying to provision my PAP2Ts to use a SVR lookup to find the Asterisk server.Im using a provisioning file that contains an element like: _sip._udp.example.com However, the PAP doesnt seem to be able to find my server with this hostname. ..
all i have a scenario where i have 2 DSL lines (i know its not that reliable but it fits the bill) connected to 1 box and would like my isp to round robin between both dsl (to allow for more capacity – each dsl could get me thru about 16-18 calls ..
all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UACASTERISK ..
list i have 1 server installed with asterisk centos and digium card i have installed the same configuration in another unit but in this unit there is no card installed i have created a sip trunk between the 2 servers like that in the server 1 with c..
On 06/13/2011 01:04 PM, bilal ghayyad wrote: > Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.n..
Quite simply: dont use a queue.Simply ring all phones at the same time using Dial(SIP/phone1&SIP/phone2&..) A queue will only send the first call until it is answered, then move on to the second one (I may be simplifying a bit) Mike From: email@example.com…