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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.

2 Answers 2

1

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.

0

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.

4
  • The problem persists after I tested multiple Codec Settings - seems like the codec isn't at fault. I think the third phone malfunctions because currently it's only in a test set-up without actual SIP Provider, just an elmeg 130j PBX. Access to the "Location" (SIP network? Confusing Web Interface...) is set to unrestricted, only then the phone is able to dial a number at all. I think it would work, if only I was registered at a proper SIP provider?
    – NoMad
    Oct 15, 2015 at 9:50
  • Ooh, I misunderstood your setup, you could configure the phone to directly call the other phones by IP, but other than that no you can't call outside numbers without a carrier Oct 16, 2015 at 6:54
  • I'm trying to call internal numbers(20, 21), configured in the elmeg 130j PBX. My 2 elmeg system phones work fine, but the Grandstream phone boots up and can't do anything. Does it need a "proper" SIP registration from a provider? I can't even dial a number before I change any of the SIP registration setting on its Web UI (Account is deactivated). Nothing is connected to the internet so far, internal use only.
    – NoMad
    Oct 16, 2015 at 10:53
  • You need a pbx/sip provider, or you can do DIRECT IP, so add a speed dial number, sat ext@phoneIP Oct 16, 2015 at 12:52

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