[asterisk-users] Sip trunk mystery

[Steve Totaro]

On Tue, Feb 26, 2008 at 12:31 PM, Dirk Enrique Seiffert
<ds at caribenet.com> wrote:
> Hello,
>
>  I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server.
>  The system is in production with local extensions, a zap trunk and a
>  working sip trunk with sipgate.de.
>
>  My asterisk server is behind a NAT/Firewall, anyhow it registers and works
>  well with sipgate.de on incoming and outgoing calls.
>
>  I aquired an account with a reseller net-voz.com: I did some testing with
>  the account directly from a Snom300 phone - works without a problem,
>  (behind the nat) I spent hours testing and adjusting the trunk
>  configuration for net-voz, maybe sombody on the list can give a helpful hint:
>
>  First of all: Registry works!
>
>  pbx*CLI> sip show registry
>  Host                            Username       Refresh State
>   Reg.Time
>  sip.net-voz.com:5060            xxxxxx6168         585 Registered
>   Tue, 26 Feb 2008 10:47:58
>  sipgate.de:5060                 xxxx0823            105 Registered
>   Tue, 26 Feb 2008 10:56:22
>
>  This is my config:
>
>  [ringtime]
>  username=5515816168
>  type=peer
>  type=friend
>  secret=118873
>  insecure=very
>  host=sip.net-voz.com
>  fromuser=5515816168
>  fromdomain=sip.net-voz.com
>  canreinvite=no
>  call-limit=50
>
>  I tried faking the user agent (without success)
>
>  useragent = Grandstream BT100 1.0.4.49
>  externip=xx.xx.116.229
>  localnet=192.168.8.0/255.255.255.0
>
>  On my gateway I can see the following with tcpdump:
>
>  listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
>  11:05:57.386827 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
>  810
>  11:05:57.452414 IP 190.144.151.212.sip > pbx.lintec.sip: SIP, length: 442
>  11:05:57.453021 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
>  385
>  11:05:57.453587 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
>  11:05:58.452868 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
>  11:06:01.453814 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
>
>  On the astersik CLI the logs show:
>
>  Audio is at 192.168.8.3 port 14800
>  Adding codec 0x4 (ulaw) to SDP
>  Adding codec 0x8 (alaw) to SDP
>  Adding non-codec 0x1 (telephone-event) to SDP
>  Reliably Transmitting (no NAT) to 190.144.151.212:5060:
>  INVITE sip:5756646022 at sip.net-voz.com SIP/2.0
>  Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
>  From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5
>  To: <sip:5756646022 at sip.net-voz.com>
>  Contact: <sip:5515816168 at 192.168.8.3>
>  Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com
>  CSeq: 103 INVITE
>  User-Agent: Asterisk PBX
>  Max-Forwards: 70
>  Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
>  algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com",
>  nonce="120404195526111105702055508208",
>  response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
>  Date: Tue, 26 Feb 2008 16:09:09 GMT
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Content-Type: application/sdp
>  Content-Length: 260
>
>  v=0
>  o=root 2381 2382 IN IP4 192.168.8.3
>  s=session
>  c=IN IP4 192.168.8.3
>  t=0 0
>  m=audio 14800 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 - - - -
>  a=ptime:20
>  a=sendrecv
>
>  ---
>  Retransmitting #1 (no NAT) to 190.144.151.212:5060:
>  INVITE sip:5756646022 at sip.net-voz.com SIP/2.0
>  Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
>  From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5
>  To: <sip:xxxxxx6022 at sip.net-voz.com>
>  Contact: <sip:xxxxx6168 at 192.168.8.3>
>  Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com
>  CSeq: 103 INVITE
>  User-Agent: Asterisk PBX
>  Max-Forwards: 70
>  Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
>  algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com",
>  nonce="120404195526111105702055508208",
>  response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
>  Date: Tue, 26 Feb 2008 16:09:09 GMT
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Content-Type: application/sdp
>  Content-Length: 260
>
>
>  It looks like the comuunication starts but then gets lost.??
>
>  Any idea is appreciated.
>
>  Thanks
>
>  Enrique
>
>
>
>  Cartagena - Colombia
>  http://www.sipcolombia.com

Does it retransmit the invite six times and then hangup?  When I have
seen this it was a firewall issue on the remote (provider) side.

Thanks,
Steve Totaro