Chan_sip / chan_pjsip error

Asterisk Asterisk SIP
1

Hello ALL,

Have strange problem. My extension couldn’t get service from Asterisk. Have next message:

NOTICE[4141]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from ‘“101” sip:101@192.168.0.15’ failed for ‘192.168.0.5:51775’ (callid: 81564ZmVhNDZkZjg1YzA1Nzc4MjZmODdmMmFkZWE0MDMxMGU) - No matching endpoint found

Where 101 is my internal number (extension)
Also have:
res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from ‘“5298617” sip:5298617@10.238.7.130’ failed for ‘202.10.4.169:5060’ (callid: 85a1871995759c62745feecb552901488a3783ed) - No matching endpoint found

Which is my trunk.

However I have this number under chan_sip driver
This is from DB
MariaDB [asterisk]> select * from sip where id=101;
±----±-----------------±----------------±------+
| id | keyword | data | flags |
±----±-----------------±----------------±------+
| 101 | pickupgroup | 1 | 23 |
| 101 | disallow | | 24 |
| 101 | allow | | 25 |
| 101 | icesupport | no | 20 |
| 101 | encryption | no | 21 |
| 101 | callgroup | 1 | 22 |
| 101 | avpf | no | 18 |
| 101 | force_avp | no | 19 |
| 101 | sendrpid | no | 11 |
| 101 | type | friend | 12 |
| 101 | nat | no | 13 |
| 101 | port | 5060 | 14 |
| 101 | qualify | yes | 15 |
| 101 | qualifyfreq | 60 | 16 |
| 101 | transport | udp,tcp,tls | 17 |
| 101 | dial | SIP/101 | 26 |
| 101 | sipdriver | chan_sip | 2 |
| 101 | secret_origional | secret123 | 3 |
| 101 | secret | secret123 | 4 |
| 101 | dtmfmode | rfc2833 | 5 |
| 101 | canreinvite | no | 6 |
| 101 | context | from-internal | 7 |
| 101 | host | dynamic | 8 |
| 101 | trustrpid | yes | 9 |
| 101 | mediaencryption | no | 10 |
| 101 | callerid | device <101> | 31 |
| 101 | account | 101 | 30 |
| 101 | permit | 0.0.0.0/0.0.0.0 | 29 |
| 101 | deny | 0.0.0.0/0.0.0.0 | 28 |
| 101 | accountcode | | 27 |
±----±-----------------±----------------±------+

less ./sip_additional.conf

[101]
deny=0.0.0.0/0.0.0.0
secret=secret123
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
dial=SIP/101
permit=0.0.0.0/0.0.0.0
callerid=Second <101>
callcounter=yes
faxdetect=no

And there is nothing into pjsip configs
[root@voipserv Thu Jan 12 18:01:38 asterisk]# grep -Hi ‘101’ ./pjsip
[root@voipserv Thu Jan 12 18:02:04 asterisk]# pwd
/etc/asterisk
[root@voipserv Thu Jan 12 18:02:08 asterisk]#

It was pjsip before my upgrade, but I decided to com back to chan_sip and now I have this. Looks like system still decides there is pjsip.

Do you have ideas, please?

Thank you in advance,
Ivan

2

Just unload all the pjsip modules if you don’t need them - manually using module unload or using noload in /etc/asterisk/modules.conf

Or remove all config (transports) from pjsip.conf so it doesn’t listen on 5060.

3

Are you using FreePBX? If so you should use their methods for disabling PJSIP or changing its port. As it is chan_pjsip is configured to listen on port 5060 probably, which is where you device is contacting.

4

Thanks a lot guys! I put noload info in modules.conf and it solve the problem, but I have another one.

When I do:

voipservCLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
voipservCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
0 active SIP channels
voipservCLI> quit
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@voipserv Mon Jan 16 20:23:07 ~]# netstat -tnupl
udp 0 0 0.0.0.0:4520 0.0.0.0: 1809/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 1809/asterisk
udp 0 0 0.0.0.0:2727 0.0.0.0:* 1809/asterisk
udp 0 0 0.0.0.0:5000 0.0.0.0:* 1809/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 1809/asterisk

but in my sip_additional.conf file I have next:

[root@voipserv Mon Jan 16 20:23:12 ~]# egrep -v ‘^;|^#|^$’ /etc/asterisk/sip.conf
[general]
include sip_additional.conf
[root@voipserv Mon Jan 16 20:40:22 ~]# less /etc/asterisk/sip_additional.conf

[101]
deny=0.0.0.0/0.0.0.0
secret=secret123
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
dial=SIP/101
permit=0.0.0.0/0.0.0.0
callerid=Second <101>
callcounter=yes
faxdetect=no

[TPG]
disallow=all
host=111.10.4.123
type=friend
port=5060
insecure=invite,port
allow=alaw
context=from-trunk
qualify=yes
directmedia=no

So,as far as I understand I should have at least one user and one trunk but I see nothing and I have no idea why. Could you give me a hint, please? At least at it should be…

Thank you in advance,
Ivan

5

You won’t have anything in the registry unless you have either a register line or a callback extension.

I’m not sure what you mean by a trunk, as it is not an Asterisk concept.

For most configurations, everything should be type=peer. type=friend can cause misoperation under some circumstances, and can cause security problems.

You are being inconsistent with canreivinvite and directmedia. The latter is the preferred name.