Hello,
I’m managing an Asterisk PBX with a simple SIP setup with three terminals peered and a trunk provided by Vodafone, working as expected with the majority of external private exchanges, but having a problem with those ISDN-based I think, please see the actual call network capture on UDP port 5060 of the Debian stretch box.
The server announces the functions and audio codecs supported to the proxy counterpart:
12:17:18.414546 IP (tos 0x0, ttl 64, id 13131, offset 0, flags [none], proto UDP (17), length 963)
46.27.XXX.XXX.5060 > 217.130.XXX.XXX.5095: [udp sum ok] SIP, length: 935
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.130.XXX.XXX:5095;branch=z9hG4bKgith79002gu5hgkacea0sb0000g00.1;received=217.130.XXX.XXX
From: <sip:902XXXXXX@remote.sip.host:5095>;tag=cb32302c
To: "lab" <sip:+34918XXXXXX@remote.sip.host:5095>;tag=as669d9e3d
Call-ID: 1e90cd473254d7d42463452d0fd5168e@remote.sip.host
CSeq: 1 INVITE
Server: Vodafone-H-500-s/v3.4.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 10800;refresher=uas
Contact: <sip:+34918XXXXXX@46.27.XXX.XXX:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 301
v=0
o=root 521990630 521990630 IN IP4 46.27.XXX.XXX
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 46.27.XXX.XXX
t=0 0
m=audio 10162 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
The proxy acknowledges
12:17:18.440061 IP (tos 0x0, ttl 125, id 0, offset 0, flags [none], proto UDP (17), length 387)
217.130.XXX.XXX.5095 > 46.27.XXX.XXX.5060: [udp sum ok] SIP, length: 359
ACK sip:+34918XXXXXX@46.27.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 217.130.XXX.XXX:5095;branch=z9hG4bKf3je61001glueqrtsmm0.1
To: "lab" <sip:+34918XXXXXX@remote.sip.host:5095>;tag=as669d9e3d
From: <sip:902XXXXXX@remote.sip.host:5095>;tag=cb32302c
Call-ID: 1e90cd473254d7d42463452d0fd5168e@remote.sip.host
CSeq: 1 ACK
Max-Forwards: 68
Content-Length: 0
And immediately hangs up
12:17:18.446047 IP (tos 0x0, ttl 125, id 0, offset 0, flags [none], proto UDP (17), length 567)
217.130.XXX.XXX.5095 > 46.27.XXX.XXX.5060: [udp sum ok] SIP, length: 539
BYE sip:+34918XXXXXX@46.27.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 217.130.XXX.XXX:5095;branch=z9hG4bKgith79002gu5hgkacea0sd0000010.1
To: "lab" <sip:+34918XXXXXX@remote.sip.host:5095>;tag=as669d9e3d
From: <sip:902XXXXXX@remote.sip.host:5095>;tag=cb32302c
Call-ID: 1e90cd473254d7d42463452d0fd5168e@remote.sip.host
CSeq: 2 BYE
Max-Forwards: 68
Content-Length: 0
Reason: Q.850;cause=111
P-Charging-Vector: icid-value=5d95ba81040bd15680d68d79fab71621
P-Charging-Function-Addresses: ccf="aaa://mm.remote.sip.host:3868;transport=tcp"
with a Q.850 status code 111, which means Protocol error, unspecified.
This happens in the very moment when the remote exchange executes divert after a keypad option. 111 is not much explicative of the problem…
Please note that the user agent is faked. I run Asterisk version 13.14.1
I would blame the destination PBX, but a SIP VoIP connection managed directly by the Vodafone router works well against the same phones…
Thanks in advance
Jordi