Hello together,
I have a Raspberry Pi with Asterisk installed.
Incomming works, outgoing not.
This are my config:
sip.conf:
[code][general]
maxexpirey=1800
defaultexpirey=120
useragent=irgendwas
externip=80.218.xxx.xxx
disallow=all
allow=ulaw
encryption=no : turns on SRTP, if you have set this then the SIP device(s) MUST use it, it’s either on or off, not optional
register => 4144xxxxxxxx:Passwort@pro2.voipgateway.org/4144xxxxxxx
[4144xxxxxxx] ; sipcall
defaultuser=4144xxxxxx
;callerid=4144xxxxxxxx <414xxxxxxx>
type=peer
remotesecret=1234
qualify=yes
;insecure=very
host=pro2.voipgateway.org
fromuser=4144xxxxxxx
fromdomain=pro2.voipgateway.org
context=meine-telefone
directmedia=no
dtmfmode=info
host=dynamic[/code]
extensions.conf:
[code][default]
include => meine-telefone
[meine-telefone]
exten => 4144xxxx,1,Dial(SIP/4144xxxxxx)
exten => _0.,1,Dial(SIP/${EXTEN:1}@4144xxxxxx,45,r)
Thats all what i don by the config files.
[/code]
Now the set debug on tell me:
<--- SIP read from UDP:192.168.61.21:2058 --->
INVITE sip:00xxxxxxxxx@192.168.61.91;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;rport
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:4144xxxxxxx@192.168.61.21:2058;line=bmtbs7h7>;reg-id=1
X-Serialnumber: 000413250195
P-Key-Flags: keys="3"
User-Agent: snom300/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 190
v=0
o=root 1514551418 1514551418 IN IP4 192.168.61.21
s=call
c=IN IP4 192.168.61.21
t=0 0
m=audio 54622 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
<------------->
--- (19 headers 10 lines) ---
Sending to 192.168.61.21:2058 (NAT)
Using INVITE request as basis request - 53cbeafbbeb0-2gmfp06nvu5q
Found peer '4144xxxxxxx' for '4144xxxxxxx' from 192.168.61.21:2058
Found RTP audio format 0
Found RTP audio format 8
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.61.21:54622
Looking for 00xxxxxxxxx in meine-telefone (domain 192.168.61.91)
list_route: hop: <sip:4144xxxxxxx@192.168.61.21:2058;line=bmtbs7h7>
<--- Transmitting (NAT) to 192.168.61.21:2058 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;received=192.168.61.21;rport=2058
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 INVITE
Server: irgendwas
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:00xxxxxxxxx@192.168.61.91:5060>
Content-Length: 0
<------------>
Audio is at 14024
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Reliably Transmitting (NAT) to 192.168.61.21:2058:
INVITE sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
Max-Forwards: 70
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Contact: <sip:4144xxxxxxx@192.168.61.91:5060>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 INVITE
User-Agent: irgendwas
Date: Sun, 20 Jul 2014 16:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 1167233314 1167233314 IN IP4 192.168.61.91
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 192.168.61.91
t=0 0
m=audio 14024 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
---
<--- Transmitting (NAT) to 192.168.61.21:2058 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;received=192.168.61.21;rport=2058
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>;tag=as49465220
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 INVITE
Server: irgendwas
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:00xxxxxxxxx@192.168.61.91:5060>
Content-Length: 0
<------------>
Retransmitting #1 (NAT) to 192.168.61.21:2058:
INVITE sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
Max-Forwards: 70
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Contact: <sip:4144xxxxxxx@192.168.61.91:5060>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 INVITE
User-Agent: irgendwas
Date: Sun, 20 Jul 2014 16:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 1167233314 1167233314 IN IP4 192.168.61.91
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 192.168.61.91
t=0 0
m=audio 14024 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
---
Retransmitting #2 (NAT) to 192.168.61.21:2058:
INVITE sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
Max-Forwards: 70
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Contact: <sip:4144xxxxxxx@192.168.61.91:5060>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 INVITE
User-Agent: irgendwas
Date: Sun, 20 Jul 2014 16:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 1167233314 1167233314 IN IP4 192.168.61.91
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 192.168.61.91
t=0 0
m=audio 14024 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.61.21:2058 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport=5060
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 INVITE
User-Agent: snom300/8.7.3.25
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:0xxxxxxxxx@pro2.voipgateway.org> for address/port to send to
set_destination: set destination to 212.117.203.44:5060
Transmitting (NAT) to 192.168.61.21:2058:
ACK sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
Max-Forwards: 70
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Contact: <sip:4144xxxxxxx@192.168.61.91:5060>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 ACK
User-Agent: irgendwas
Content-Length: 0
---
Scheduling destruction of SIP dialog '7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org' in 6400 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.61.21:2058 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;received=192.168.61.21;rport=2058
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>;tag=as49465220
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 INVITE
Server: irgendwas
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0
<------------>
Retransmitting #1 (NAT) to 192.168.61.21:2058:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;received=192.168.61.21;rport=2058
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>;tag=as49465220
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 INVITE
Server: irgendwas
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0
---
<--- SIP read from UDP:192.168.61.21:2058 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport=5060
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 INVITE
User-Agent: snom300/8.7.3.25
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.61.21:2058:
ACK sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
Max-Forwards: 70
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Contact: <sip:4144xxxxxxx@192.168.61.91:5060>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 ACK
User-Agent: irgendwas
Content-Length: 0
---
<--- SIP read from UDP:192.168.61.21:2058 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport=5060
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 INVITE
User-Agent: snom300/8.7.3.25
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.61.21:2058:
ACK sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
Max-Forwards: 70
From: "4144xxxxxxx" <sip:4144xxxxxxx@pro2.voipgateway.org>;tag=as2acfca9c
To: <sip:0xxxxxxxxx@pro2.voipgateway.org>
Contact: <sip:4144xxxxxxx@192.168.61.91:5060>
Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
CSeq: 102 ACK
User-Agent: irgendwas
Content-Length: 0
---
<--- SIP read from UDP:192.168.61.21:2058 --->
ACK sip:00xxxxxxxxx@192.168.61.91;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;rport
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>;tag=as49465220
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:4144xxxxxxx@192.168.61.21:2058;line=bmtbs7h7>;reg-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '53cbeafbbeb0-2gmfp06nvu5q' Method: ACK
<--- SIP read from UDP:192.168.61.21:2058 --->
ACK sip:00xxxxxxxxx@192.168.61.91;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;rport
From: "4144xxxxxxx" <sip:4144xxxxxxx@192.168.61.91>;tag=dlfj53tk7d
To: <sip:00xxxxxxxxx@192.168.61.91;user=phone>;tag=as49465220
Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:4144xxxxxxx@192.168.61.21:2058;line=bmtbs7h7>;reg-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
raspberrypi*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
root@raspberrypi:~#i dont found any foulds :s
can you help me please