Author Topic: VoIp rpoble using another provider  (Read 624 times)

mitsus

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VoIp rpoble using another provider
« on: October 30, 2007, 12:16:11 pm »
Hi all,
i've a problem when i try to do and receive a call via an external sip provider NOVERCA.
I'm able to send a call to Linux MCE softPhone (it ringing), but i the RTP flow don't work.
Below i post my asterisk log
Code: [Select]
<-- SIP read from 192.168.80.1:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK37743f71;rport=5060
From: "0039066114" <sip:0039066114@192.168.80.1>;tag=as696bf54c
To: <sip:2222@192.168.80.1:5061>;tag=1954321300
Call-ID: 5f2980e529469ad30927cca1661ff15e@192.168.80.1
CSeq: 102 INVITE
Contact: <sip:2222@192.168.80.1:5061>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length:   213

v=0
o=root 123456 654321 IN IP4 192.168.1.10
s=A conversation
c=IN IP4 192.168.1.10
t=0 0
m=audio 7078 RTP/AVP 0 8 101
b=AS:-1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

--- (10 headers 10 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.10:7078
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:2222@192.168.80.1:5061>
set_destination: Parsing <sip:2222@192.168.80.1:5061> for address/port to send to
set_destination: set destination to 192.168.80.1, port 5061
Transmitting (NAT) to 192.168.80.1:5061:
ACK sip:2222@192.168.80.1:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK2566d099;rport
From: "0039066114" <sip:0039066114@192.168.80.1>;tag=as696bf54c
To: <sip:2222@192.168.80.1:5061>;tag=1954321300
Contact: <sip:0039066114@192.168.80.1>
Call-ID: 5f2980e529469ad30927cca1661ff15e@192.168.80.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/2222-6cbd answered SIP/linux_mce1-c5d8
We're at 192.168.1.10 port 17796
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.217.147.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.217.147.203:5060;branch=z9hG4bKa6d43d11db3e00edec5f0f29;received=213.217.147.203;rport=5060
Via: SIP/2.0/UDP  213.217.147.248:5060;branch=z9hG4bKE326F3
Record-Route: <sip:213.217.147.203;lr>
From: <sip:0039066114@213.217.147.248>;tag=3DCD25E0-23AF
To: <sip:00390661244130@213.217.147.203>;tag=as51aff481
Call-ID: 6F5DEE67-67A011D5-834DB747-8AF6A98@213.217.147.248
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2222@192.168.1.10>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 7297 7298 IN IP4 192.168.1.10
s=session
c=IN IP4 192.168.1.10
t=0 0
m=audio 17796 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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 - - - -

--- (8 headers 0 lines)---
dcerouter*CLI>
<-- SIP read from 192.168.80.1:5061:
SIP/2.0 481 Subcription Does Not Exist
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6e0a61cf;rport=5060
From: "asterisk" <sip:asterisk@192.168.80.1>;tag=as55eec798
To: <sip:2222@192.168.80.1:5061>;tag=2040230086
Call-ID: 067874834dc3ee461571a4be3daedf7f@192.168.80.1
CSeq: 102 NOTIFY
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0


--- (8 headers 0 lines)---
    -- Got SIP response 481 "Subcription Does Not Exist" back from 192.168.80.1
Destroying call '067874834dc3ee461571a4be3daedf7f@192.168.80.1'
Destroying call '6101a3cc58f8e4ba52dc48de6b06e759@192.168.80.1'
Destroying call '1348218129@192.168.80.1'

Tell me if i must post others information, about for example my sip configuration files.

Thank you