I'm installing a SIP Phone in a VoIP environment. There are 2 System Phones which work flawlessly (same manufacturer as PBX), and the third phone can be called, but it can't call the other 2 phones.
The PBX shows an error: "No matching Codecs! Call rejected" This is the conversation from the third phone's perspective:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport
From: <sip:192.168.0.250>;tag=1540961770
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 160 INVITE
Contact: <sip:192.168.0.14:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.5.18
Privacy: none
P-Preferred-Identity: <sip:192.168.0.250>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 306
v=0
o=- 8000 8000 IN IP4 192.168.0.14
s=SIP Call
c=IN IP4 192.168.0.14
t=0 0
m=audio 5004 RTP/AVP 9 8 18 2 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport=5060
From: <sip:192.168.0.250>;tag=1540961770
To: <sip:[email protected]>;tag=74C1BCB3775433109F0E49A014240025
Call-ID: [email protected]
CSeq: 160 INVITE
Content-Length: 0
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport
From: <sip:192.168.0.250>;tag=1540961770
To: <sip:[email protected]>;tag=74C1BCB3775433109F0E49A014240025
Call-ID: [email protected]
CSeq: 160 ACK
Content-Length: 0
But from an incoming Call:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060;transport=udp>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: hybird_130j V.9.1 Rev. 10 (Patch 4) IPSec
Alert-Info: <http://127.0.0.1>;info=alert-internal
Allow-Events: refer, message-summary, dialog
P-Asserted-Identity: "Sys Tel 20" <sip:[email protected];user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 328
v=0
o=- 71 1 IN IP4 192.168.0.250
s=SIP call
c=IN IP4 192.168.0.250
t=0 0
m=audio 10848 RTP/AVP 0 8 18 2 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.5.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>;tag=471383942
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:192.168.0.14:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.5.18
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>;tag=471383942
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:192.168.0.14:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.5.18
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 306
Note that both offer PCMU/PCMA and some other codecs. Why does the call fail?
Phones' IPs are 192.168.0.12 and 192.168.0.14, the PBX has 192.168.0.250.
The sad truth is, those non-elmeg system phones didn't register correctly. I had to assign PINs for the Users in the Hybird 130j Interface, which is then used to authenticate with the System.
Elmeg didn't bother to state anywhere that this PIN must be set and is used as the SIP password for non-elmeg phones.
Remove g729 unless you have the license.
Use 711a, 711u, 722, GSM.
Check that the phones are all registering, and that they are using the same transport (udp) as well as nat settings are all equal.