Hello
I have configured a sip-trunk towards a trunk-operator here in Sweden. I have two problems with the trunk.
1. I can register without any problem but when thirdlane pbx sends the keep a-live to the trunk-operator it looks like this:
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
OPTIONS sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0887d5a2;rport
From: "unknown" ;tag=as336fe443
To:
Contact:
Call-ID: 07b2da0e2628e71c62967d1068a3fc93@xxx.xxx.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Mar 2010 10:33:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
i guess i have missed out on something since the registrastration looks like this:
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78970ee4;rport
From: ;tag=as1ee4edc5
To:
Call-ID: 2a95c19b55bfa67a19deb2f454ddea31@127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="username", realm="demoasterix.se", algorithm=MD5, uri="sip:xxx.xxx.xxx.xxx", nonce="1267525971:78eb8475eb2c8a0c7a1ee5df5fc830ed", response="4ac6babf253f9cf597d295894089ab35", qop=auth, cnonce="26be73de", nc=00000001
Expires: 120
Contact:
Event: registration
Content-Length: 0
2. When i call in on the same trunk it seems like thirdlane doesnt handle the codec in 200 ok on the invite correctly. here is the invite and the 200 ok:
<--- SIP read from xxx.xxx.xxx.xxx:5060 --->
INVITE sip:0855776043@demoasterix.se;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-d8754z-180e4755c278507d-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "0855776043"
From: "0852281806";tag=cc1a9520
Call-ID: OThjZjJlNWFhNDA2OTM0YzVlNmQxNTg4MjExYjlkYzk.
CSeq: 1 INVITE
Allow: INVITE, ACK, BYE, CANCEL
Content-Type: application/sdp
User-Agent: LEICA-1.8.30-RC2
X-Ecan: On
Content-Length: 271
v=0
o=- 1852272920 0 IN IP4 xxx.xxx.xxx.xxx
s=-
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 4070 RTP/AVP 8 0 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=cpar:T38FaxVersion:0
a=cpar:T38MaxBitRate:14400
a=sendrecv
m=image 4072 udptl t38
200 ok:
<--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-d8754z-180e4755c278507d-1---d8754z-;received=xxx.xxx.xxx.xxx;rport=5060
From: "0852281806";tag=cc1a9520
To: "0855776043";tag=as4ef35716
Call-ID: OThjZjJlNWFhNDA2OTM0YzVlNmQxNTg4MjExYjlkYzk.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 3444 3444 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=image 4490 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC
as far as i can see the media description in the 200 ok are missing.
Here is a link with sip.conf, registration and the call. Hope I can get some help.
http://www.pastebin.org/99873
Br
Jörgen
Hello 1. yes but it seems
Hello
1. yes but it seems like i have missed out on some configuration since the codec from thirdlane is wrong or nothing media description.
2. That is a regular call.
3. Its turned off now but its still the same problem.
Br
Jörgen
so explain to me what happens
so explain to me what happens when you place a call to your carrier?
when you receive a call from your carrier?
does asterisk dialplan activate at all? if so where does it fail?
Hello When im calling the
Hello
When im calling the number from my cellphone i see that the call get to the extension. since no device is connected to the extension it goes into vm. all that is fine. if i do a tcpdump on the server i can see the call perfectly but i can only hear audio from the thirdlane. not anything to it and that is because the 200 ok from the invite are not correct.
i can see in the thirdlane cli that the voicemail activates i see that this plays -- Playing '/var/spool/asterisk/voicemail/default-iptelefonibolaget/1103/unavail' (language 'en')
the audio is missing :) and there are no firewall infront of the server.
/Jörgen
so your problem is 1-way
so your problem is 1-way audio?
I want you to configure a softphone and do a test where you talk to the caller, verify that you are in fact having a 1-way audio issue.
I want you to place a call to your cell from the soft phone and again confirm the same 1-way audio issue.
In both cases is it the audio stream from the provider to asterisk that fails?
the fact that you get audio in the other direction means that SDP completed without error.
Hello The problem is that
Hello
The problem is that there is no audio at all and that the call is not directed to the device. If i change the number to a different trunk-operator everything works perfect. if i look at inbound roots it looks fine.
here is a new logfile from asterisk when i call in.
http://www.pastebin.org/100096
the 200 ok are missing some codecs again and it seems to be something wrong at line 198.
i have to add that it works perfectly to call 1103 extension from another extension in the same tenant.
br
jörgen
well yea.. its not like
well yea.. its not like chan_sip just doesnt work, i mean 'cmon how do you think asterisk got so popular if it couldnt negotiate sip. More than likely you're experiencing some sort of inter-op problem between one type of switch and asterisk. If you genuinely are experiencing a sip interop issue theres nothing we can do to help you. All of sip negotiation occurs within chan_sip. If theres a problem with dialog between two switches you'll most likely have to purchase a support incident with digium to get it resolved via code.
Hello I manage to solve the
Hello
I manage to solve the problem by change t38pt_udptl=yes to t38pt_udptl=false in the sip..conf.
Now it works receiving calls, call out and receive fax.
Im using Asterisk version 1.4
Br
Jörgen
i thought you said it wasn't
i thought you said it wasn't a fax call? your setting just prevents negotiation of T38 fax but should not impact a regular call. Are you saying your provider sucks so bad that every call negotiates T38 even when there is no fax? wow that sucks. Glad to see you got it working.
Eeman yes it was no fax but
Eeman yes it was no fax but we got that info anyway i the sip header.
Im glad i made it working
Thanks for you help anyway eeman.
/Jörgen
not sure I am following you and what exactly isnt working..
1. SIP does not do anything with codec negotiation.. codec negotiation and port negotiation occurs in SDP after SIP has negotiated the call.
2. This looks like a T38 fax call, are you saying regular calls work but T38 does not?
3. why do you have NAT turned on for this trunk? if you are both running on public IP's then you should set nat=no.