Helo,
i am using asterisk 1.4 with freepbx. behind nat.
port 5060 as weel as configured rtp ports on asterisk a forwarded to asterisk
I am using a sip provider to make voip calls.
I am calling a phone which is connected to a ISDN PBX. We can talk and everythink seems to be ok. Now the other end is tranfering the call to a other phone and Iam losing the media stream. I can hear for on second the musicohnhold from asterik ( not from the other pbx) and it is quit.
When the other end picks up the phone where the call was transfered it can`t also hear me.
First I was thinking asterisk is losing the rtp stream, but as I mad a tcpdump on asterisk, I found out the transfered call from the other end goes to my asterisk and from there not to my local sip phone.
her a dump from asterisk cli:
INVITE sip:03445xxxxxx@strato-iphone.de SIP/2.0
Via: SIP/2.0/UDP 194.97.176.xxx:5060;branch=z9hG4bK7170fe46;rport
From: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
To: <sip:03445xxxxxx@strato-iphone.de>
Contact: <sip:USERNAME@194.97.176.xxx>
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="USERNAME", realm="strato-iphone.de", algorithm=MD5, uri="sip:03445xxxxxx@strato-iphone.de", nonce="471488614bf8950a611254bd02bd2d5970b7d62e", response="bdd80954ecdf2677f3d5df8742f8d645", opaque="", qop=auth, cnonce="242a68c1", nc=00000001
Date: Tue, 16 Oct 2007 09:41:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 3303 3304 IN IP4 194.97.176.xxx
s=session
c=IN IP4 194.97.176.xxx
t=0 0
m=audio 10094 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<-- SIP read from 194.97.40.217:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 194.97.176.xxx:5060;branch=z9hG4bK7170fe46;rport=5060
From: "271xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
To: <sip:03445xxxxxx@strato-iphone.de>
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 103 INVITE
Content-Length: 0
<-- SIP read from 194.97.40.217:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 194.97.176.xxx:5060;branch=z9hG4bK7170fe46;rport=5060
From: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
To: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 103 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length: 191
v=0
o=- 459148704 459148704 IN IP4 194.97.57.196
s=session
c=IN IP4 194.97.57.196
t=0 0
m=audio 32740 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (9 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 194.97.57.196:32740
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/STRATO-1-082905e8 is ringing
-- SIP/STRATO-1-082905e8 is making progress passing it to SIP/24-b6056478
<-- SIP read from 194.97.40.217:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 194.97.176.xxx:5060;branch=z9hG4bK7170fe46;rport=5060
Record-Route: <sip:194.97.40.217;ftag=as3a5f0a14;lr=on>
From: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
To: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 103 INVITE
Contact: <sip:194.97.45.167:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Supported: 100rel, timer, replaces
Content-Length: 191
v=0
o=- 459148704 459148705 IN IP4 194.97.57.196
s=session
c=IN IP4 194.97.57.196
t=0 0
m=audio 32740 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (13 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 194.97.57.196:32740
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Transmitting (no NAT) to 194.97.40.217:5060:
ACK sip:194.97.45.167:5060 SIP/2.0
Via: SIP/2.0/UDP 194.97.176.xxx:5060;branch=z9hG4bK60f45fc0;rport
Route: <sip:194.97.40.217;ftag=as3a5f0a14;lr=on>
From: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
To: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
Contact: <sip:USERNAME@194.97.176.xxx>
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<-- SIP read from 194.97.40.217:5060:
INVITE sip:USERNAME@194.97.176.xxx SIP/2.0
Record-Route: <sip:194.97.40.217;ftag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD;lr=on>
Via: SIP/2.0/UDP 194.97.40.217;branch=z9hG4bKb9ee.489a8625.0
Via: SIP/2.0/UDP 194.97.45.167:5060;branch=z9hG4bK000423D241B64ACC5E2B9EA75099
From: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
To: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 7237 INVITE
Contact: <sip:03445xxxxxx@194.97.45.167:5060>
Max-Forwards: 16
Content-Type: application/sdp
Supported: 100rel, timer, replaces
Content-Length: 191
v=0
o=- 459148704 459148706 IN IP4 194.97.57.196
s=session
c=IN IP4 194.97.57.196
t=0 0
m=audio 32740 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendonly
--- (13 headers 9 lines) ---
Using INVITE request as basis request - 49de9426035c645e41ca699444b42ab5@strato-iphone.de
Sending to 194.97.40.217 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 194.97.57.196:32740
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- Music class default requested but no musiconhold loaded.
We're at 194.97.176.xxx port 10094
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 194.97.40.217:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.97.40.217;branch=z9hG4bKb9ee.489a8625.0;received=194.97.40.217
Via: SIP/2.0/UDP 194.97.45.167:5060;branch=z9hG4bK000423D241B64ACC5E2B9EA75099
Record-Route: <sip:194.97.40.217;ftag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD;lr=on>
From: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
To: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 7237 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:USERNAME@194.97.176.xxx>
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 3303 3305 IN IP4 194.97.176.xxx
s=session
c=IN IP4 194.97.176.xxx
t=0 0
m=audio 10094 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<-- SIP read from 194.97.40.217:5060:
ACK sip:USERNAME@194.97.176.xxx SIP/2.0
Record-Route: <sip:194.97.40.217;ftag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD;lr=on>
Via: SIP/2.0/UDP 194.97.40.217;branch=0
Via: SIP/2.0/UDP 194.97.45.167:5060;branch=z9hG4bK000423D241B64ACC5E32B7221F5E
From: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
To: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 7237 ACK
Contact: <sip:03445xxxxxx@194.97.45.167:5060>
Max-Forwards: 16
Content-Length: 0
<-- SIP read from 194.97.40.217:5060:
BYE sip:USERNAME@194.97.176.xxx SIP/2.0
Record-Route: <sip:194.97.40.217;ftag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD;lr=on>
Via: SIP/2.0/UDP 194.97.40.217;branch=z9hG4bK9aee.f39a1374.0
Via: SIP/2.0/UDP 194.97.45.167:5060;branch=z9hG4bK000423D241B64ACC65CA87B6D3C0
From: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
To: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 7238 BYE
Contact: <sip:03445xxxxxx@194.97.45.167:5060>
Max-Forwards: 16
Content-Length: 0
--- (11 headers 0 lines) ---
Sending to 194.97.40.217 : 5060 (non-NAT)
Transmitting (no NAT) to 194.97.40.217:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.97.40.217;branch=z9hG4bK9aee.f39a1374.0;received=194.97.40.217
Via: SIP/2.0/UDP 194.97.45.167:5060;branch=z9hG4bK000423D241B64ACC65CA87B6D3C0
Record-Route: <sip:194.97.40.217;ftag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD;lr=on>
From: <sip:03445xxxxxx@strato-iphone.de>;tag=3qagEqC8X20004jR0A0Ol0IFHCq0s2CsD
To: "713xxx" <sip:USERNAME@strato-iphone.de>;tag=as3a5f0a14
Call-ID: 49de9426035c645e41ca699444b42ab5@strato-iphone.de
CSeq: 7238 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:USERNAME@194.97.176.xxx>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
configuration of peer for sip provider:
sip show peer strato-1
* Name : STRATO-1
Secret : <Set>
MD5Secret : <Not set>
Context : from-pstn
Subscr.Cont. : <Not set>
Language : de
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
FromUser : USERNAME
FromDomain : strato-iphone.de
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Insecure : port,invite
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : strato-iphone.de
Addr->IP : 194.97.40.217 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: USERNAME
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw,alaw)
Status : Unmonitored
Useragent :
Reg. Contact :configuration of local sip peer:
sip show peer 24
* Name : 24
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language : de
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
Mailbox : 24@device
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Dynamic : Yes
Callerid : "device" <24>
Expire : 3164
Insecure : no
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.255.124 Port 2051
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 24
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw,alaw)
Status : OK (14 ms)
Useragent : snom360/6.5.10
Reg. Contact : sip:24@192.168.255.124:2051;line=ow5tqzkrhere is a sipdialog:
ip 192.168.255.150 local sip client
ip 192.168.255.100 asterisk
ip 194.97.40.217 strato sip proxy
konabi.de/asterisk/sipdialog.pdf
If I use other sip providers, I do not have this problem.
The wehen the called party transfer the call I hear musiconhold from the other PBX and everything works.
I am thanksfull for any hint.
Sven[/url][/code]