, I am trying to enable SIP SIMPLE communication in my test environment(Asterisk 13.6.0)I have two problems:1. Using messagesend(), I dont want my users to be able to change their own callerid name. I want the name that appears in the ${MESSAGE(fro..
Author : Julien Sansonnens
Home » Articles posted by Julien Sansonnens
Some of my users connect to my asterisk box using SIP, other using iax(in users.conf, I set hasiax=yes for those users).How do I detect which protocol some user is using ? I cannot find any variable which contains that information.Reason is: I need t..
everybody,Im experiencing quite a strange issue with some voip carriers when Itry to call my own GSM cell phone, using international route (Im in switzerland).With some budget routes (value route from voip.ms for example), I get a fast busy tone, congest..
I upgraded to Asterisk 13.6.0 compiled with pjproject 2.4.5 (dont know if its related). I see those messages in the console:[Dec6 10:26:33] ERROR[40904]: res_hep.c:517 hep_queue_cb: Error [1]while sending packet to HEPv3 server: Operation not permitt..
Does the ENUM service e164.org died? All queries that I do lead toNXDOMAIN. I feel that the base has simply been empty for several months. Maybe some old addresses are still resolved, but the newer ones are not.The website still works, but it seems t..
When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc …Is there a way to force URI calls through the PBX? I have fo..
Do you know if there are any e.164 country codes (CC) that provide a gateway from PSTN reachable numbers to URI (SIP) for their whole range?The only example I knowis iNum (+883 5100 country code) Each iNum number can be converted into an URI in the f..