Why are calls from the built-in sip client on a smartphone not going through?

Question

Android

Good afternoon community. The problem is this: asterisk 11 was raised on zyxel keenetic ultra via opkg. The dialplan is configured, several phones are connected. But the problem is that the built-in android sip client (which is configured in a regular dialer) does not want to call through the sip channel. Incoming passes, and outgoing stupidly cut off a couple of seconds after dialing. My guess is that this is because the phone is dialing in the form [email protected], where ip is the asterisk address. However, for some reason I can’t track this either, because calls are not reflected in the asterisk log, although I tried to set up verbose with level 10. Actually, there are two questions. First: is there a ready-made solution for calls via sip? Second: how can you track which phone is dialing for diagnostics?
I am a beginner user of asterisk, so if you need additional information, please write to me and I will add it.

UPD
It is noticed that messages about current calls usually appear in the command interface. When you try to call from an android client - emptiness. Does that mean it doesn't work?

Answer 1

CsipSipple and a regular client eat the battery if the option to receive incoming calls is enabled. The coolest Bria client with the push option, eats no more viber and does not fall out of memory. There is a broken version of Bria on w3bsit3-dns.com, it works fine. To answer your question, you need to look at the SIP Dump of the call.
Install sgrep and dump the call.
https://github.com/irontec/sngrep/wiki/Installing-...
sngrep -IO sip.pcap XXXXXX
where XXXXXX is the number to call.