Hallo Leute,
ich habe ein elmeg IP 290 an einer vlines Entire 2xS0 laufen mit Asterisk Version 1.2.20. Die Firmware des elmeg, das eigentlich ein snom 190 ist, ist die 3.60s
Wenn ich mit anderen SIP-Geräten vom gleichen Asterisk rauswähle tritt das Problem nicht auf, daher kann es eigentlich nur am Telefon liegen.
Hier ein SIP Debug eines ausgehenden Calls (ich habe die gewählte Nummer unkenntlich gemacht):
Vielleicht sieht ja jemand was. Danke
Matthias
EDIT der_Gersthofer: Bitte Fragen/Probleme zu Elmeg Geräten in den ELMEG Bereich
ich habe ein elmeg IP 290 an einer vlines Entire 2xS0 laufen mit Asterisk Version 1.2.20. Die Firmware des elmeg, das eigentlich ein snom 190 ist, ist die 3.60s
Wenn ich mit anderen SIP-Geräten vom gleichen Asterisk rauswähle tritt das Problem nicht auf, daher kann es eigentlich nur am Telefon liegen.
Hier ein SIP Debug eines ausgehenden Calls (ich habe die gewählte Nummer unkenntlich gemacht):
Code:
processing...
IAX2 Debugging Disabled
SIP Debugging Disabled
changing debug level for all ports to 0
active Debug peer: 41
SIP Debugging Enabled for IP: 192.168.0.61:2071
<-- SIP read from 192.168.0.61:2071:
INVITE sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-xkqk31xjmmgk;rport
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2071;line=mexm4kdk>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60x
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 368
v=0
o=root 1626148452 1626148452 IN IP4 192.168.0.61
s=call
c=IN IP4 192.168.0.61
t=0 0
m=audio 59096 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
�--- (17 headers 17 lines) ---
�Using INVITE request as basis request - 3c27977b72bf-pwxmvkd9ix9p@snom190
�Sending to 192.168.0.61 : 2071 (NAT)
�Reliably Transmitting (no NAT) to 192.168.0.61:2071:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-xkqk31xjmmgk;received=192.168.0.61;rport=2071
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>;tag=as675fd4de
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 1 INVITE
User-Agent: Vlines accessVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09e82ab7"
Content-Length: 0
---
�Scheduling destruction of call '3c27977b72bf-pwxmvkd9ix9p@snom190' in 15000 ms
�Found user '41'
<-- SIP read from 192.168.0.61:2071:
ACK sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-xkqk31xjmmgk;rport
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>;tag=as675fd4de
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2071;line=mexm4kdk>
Content-Length: 0
�--- (9 headers 0 lines) ---
�
<-- SIP read from 192.168.0.61:2071:
INVITE sip:[email protected]:5070;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-sv4656nefnvw;rport
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2071;line=mexm4kdk>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60x
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest username="41",realm="asterisk",nonce="09e82ab7",uri="sip:[email protected]:5070;user=phone",response="756e40304d2f21a63acbbe53fb8eefd2",algorithm=md5
Content-Type: application/sdp
Content-Length: 368
v=0
o=root 1626148452 1626148452 IN IP4 192.168.0.61
s=call
c=IN IP4 192.168.0.61
t=0 0
m=audio 59096 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
�--- (18 headers 17 lines) ---
�Using INVITE request as basis request - 3c27977b72bf-pwxmvkd9ix9p@snom190
�Sending to 192.168.0.61 : 2071 (NAT)
�Found user '41'
�Found RTP audio format 0
�Found RTP audio format 8
�Found RTP audio format 9
�Found RTP audio format 2
�Found RTP audio format 3
�Found RTP audio format 18
�Found RTP audio format 4
�Found RTP audio format 101
�Peer audio RTP is at port 192.168.0.61:59096
�Found description format pcmu
�Found description format pcma
�Found description format g722
�Found description format g726-32
�Found description format gsm
�Found description format g729
�Found description format g723
�Found description format telephone-event
�Capabilities: us - 0x11e (gsm|ulaw|alaw|g726|g729), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x11e (gsm|ulaw|alaw|g726|g729)
�Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
�Looking for 06081123456 in 41 (domain 192.168.0.2)
�list_route: hop: <sip:[email protected]:2071;line=mexm4kdk>
Transmitting (no NAT) to 192.168.0.61:2071:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-sv4656nefnvw;received=192.168.0.61;rport=2071
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 2 INVITE
User-Agent: Vlines accessVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]:5070>
Content-Length: 0
---
We're at 192.168.0.2 port 10078
Adding codec 0x8 (alaw) to SDP
�Adding codec 0x4 (ulaw) to SDP
�Adding codec 0x10 (g726) to SDP
�Adding codec 0x2 (gsm) to SDP
�Adding codec 0x100 (g729) to SDP
�Adding non-codec 0x1 (telephone-event) to SDP
�Reliably Transmitting (no NAT) to 192.168.0.61:2071:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-sv4656nefnvw;received=192.168.0.61;rport=2071
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>;tag=as734c1bcd
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 2 INVITE
User-Agent: Vlines accessVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]:5070>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 5339 5340 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 10078 RTP/AVP 8 0 2 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
<-- SIP read from 192.168.0.61:2071:
ACK sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61:2071;branch=z9hG4bK-dxye8qwc0gfm;rport
From: <sip:[email protected]:5070>;tag=0g9pyod1zs
To: <sip:[email protected]:5070;user=phone>;tag=as734c1bcd
Call-ID: 3c27977b72bf-pwxmvkd9ix9p@snom190
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2071;line=mexm4kdk>
Content-Length: 0
�--- (9 headers 0 lines) ---
Vielleicht sieht ja jemand was. Danke
Matthias
EDIT der_Gersthofer: Bitte Fragen/Probleme zu Elmeg Geräten in den ELMEG Bereich
Zuletzt bearbeitet: