Hey Daniel,
Unfortunately, i do not maintain the system at 80.234.135.99. However, i am not sure there is anything wrong with their system anyway.... as you see, i have attempted to dial from my UA once dialling a full address (in this case 3018 - at - sip.prodec.tv) which was responded to correctly, and once by dialling 839503018 where 83950 should be translated as sip.prodec.tv using the PDT module. Hence, ser should finally send the invite to 3018 - at - sip.prodec.tv in the same was as if i dialled the full address manually, no? There should be no difference, once that invite request, has left ser (161.30.94.136) in how the remote system handles it.
As you can see from the syslog:
ACC: transaction answered: method=INVITE, i-uri=sip:839503018 at sip.dev.inmarsat.com, o-uri=sip:3018 at sip.prodec.tv <mailto:o-uri=sip:3018@sip.prodec.tv>
ACC: transaction answered: method=INVITE, i-uri=sip:3018 at sip.prodec.tv, o-uri=sip:3018 at sip.prodec.tv <mailto:o-uri=sip:3018@sip.prodec.tv> , call_id=9C912086-C197-484D-8AD2-E261F1A3234A(a)161.30.94.150, from=Dave Bath <sip:admin at sip.dev.inmarsat.com>;tag=2122510239, code=487
for some reason using the pdt module appears to issue a malformed invite request... unless soemthing in ser.cfg is not configured correctly.
Is my logic totally off? I appreciate your words of wisdom very much...
Dave
________________________________
From: Daniel-Constantin Mierla [mailto:daniel@iptel.org]
Sent: Tue 10/08/2004 18:03
To: Dave Bath
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] PDT module
Hello,
I see that the request was forwarded from 161.30.94.136:5060 to
80.234.135.99:5060 and you get back a 100 trying. PDT does its job on
161.30.94.136, as far as I assume, so there should be a problem on the
other site 80.234.135.99, could you check that system too?
Daniel
On 8/10/2004 12:06 PM, Dave Bath wrote:
> Hey Daniel and all,
>
> Many thanks for looking into this. The first ngrep is dialling the
> full address, the second is using the pdt module.
>
> ------------------------- BEGIN dialling full external SIP address
> ---------
>
> U 161.30.94.150:5060 -> 161.30.94.136:5060
>
> INVITE sip:3018@sip.prodec.tv SIP/2.0..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport;branch=z9hG4bKBAF99F2231994F0D9A5
>
> 3836019EAC108..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=399847332..To:
> <sip:3018@sip.prodec.tv>..Conta
>
> ct: <sip:admin@161.30.94.150:5060>..Call-ID:
> FC2BDDD4-9E47-4C5A-9034-4977FC86290D@161.30.94.150..CSeq: 35761 INVIT
>
> E..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite
> release 1103m..Content-Length: 298....v=0..
>
> o=admin 434214617 434214657 IN IP4 161.30.94.150..s=X-Lite..c=IN IP4
> 161.30.94.150..t=0 0..m=audio 8000 RTP/AVP 0
>
> 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3
> gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97
>
> speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..
>
> #
>
> U 161.30.94.136:5060 -> 161.30.94.150:5060
>
> SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport=5060;branch=z9hG4bKB
>
> AF99F2231994F0D9A53836019EAC108..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=399847332..To: <sip:3018@sip
>
> .prodec.tv>..Call-ID:
> FC2BDDD4-9E47-4C5A-9034-4977FC86290D@161.30.94.150..CSeq: 35761
> INVITE..Server: Sip EXpress
>
> router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
> 161.30.94.136:5060 "Noisy feedback tells: pid=13743
>
> req_src_ip=161.30.94.150 req_src_port=5060
> in_uri=sip:3018@sip.prodec.tv out_uri=sip:3018@sip.prodec.tv via_cnt==
>
> 1"....
>
> #
>
> U 161.30.94.136:5060 -> 80.234.135.99:5060
>
> INVITE sip:3018@sip.prodec.tv SIP/2.0..Record-Route:
> <sip:3018@161.30.94.136;ftag=399847332;lr=on>..Via: SIP/2.0/U
>
> DP 161.30.94.136;branch=z9hG4bK51f6.ae27ae87.0..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport=5060;branch=z9hG4bKBAF99
>
> F2231994F0D9A53836019EAC108..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=399847332..To: <sip:3018@sip.pro
>
> dec.tv>..Contact: <sip:admin@161.30.94.150:5060>..Call-ID:
> FC2BDDD4-9E47-4C5A-9034-4977FC86290D@161.30.94.150..CSe
>
> q: 35761 INVITE..Max-Forwards: 69..Content-Type:
> application/sdp..User-Agent: X-Lite release 1103m..Content-Length
>
> : 298....v=0..o=admin 434214617 434214657 IN IP4
> 161.30.94.150..s=X-Lite..c=IN IP4 161.30.94.150..t=0 0..m=audio 8
>
> 000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8
> pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/800
>
> 0..a=rtpmap:97 speex/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-15..
>
> #
>
> U 161.30.94.136:5060 -> 80.234.135.99:5060
>
> INVITE sip:3018@sip.prodec.tv SIP/2.0..Record-Route:
> <sip:3018@161.30.94.136;ftag=399847332;lr=on>..Via: SIP/2.0/U
>
> DP 161.30.94.136;branch=z9hG4bK51f6.ae27ae87.0..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport=5060;branch=z9hG4bKBAF99
>
> F2231994F0D9A53836019EAC108..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=399847332..To: <sip:3018@sip.pro
>
> dec.tv>..Contact: <sip:admin@161.30.94.150:5060>..Call-ID:
> FC2BDDD4-9E47-4C5A-9034-4977FC86290D@161.30.94.150..CSe
>
> q: 35761 INVITE..Max-Forwards: 69..Content-Type:
> application/sdp..User-Agent: X-Lite release 1103m..Content-Length
>
> : 298....v=0..o=admin 434214617 434214657 IN IP4
> 161.30.94.150..s=X-Lite..c=IN IP4 161.30.94.150..t=0 0..m=audio 8
>
> 000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8
> pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/800
>
> 0..a=rtpmap:97 speex/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-15..
>
> #
>
> U 80.234.135.99:5060 -> 161.30.94.136:5060
>
> SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> 161.30.94.136;branch=z9hG4bK51f6.ae27ae87.0..Via: SIP/2.0/UDP
> 161.30.94.150:5
>
> 060;rport=5060;branch=z9hG4bKBAF99F2231994F0D9A53836019EAC108..From:
> "Dave Bath" <sip:admin@sip.dev.inmarsat.com>;
>
> tag=399847332..To: <sip:3018@sip.prodec.tv>..Call-ID:
> FC2BDDD4-9E47-4C5A-9034-4977FC86290D@161.30.94.150..CSeq: 35
>
> 761 INVITE..Content-Length: 0....
>
> #
>
> U 80.234.135.99:5060 -> 161.30.94.136:5060
>
> SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> 161.30.94.136;branch=z9hG4bK51f6.ae27ae87.0..Via: SIP/2.0/UDP
> 161.30.94.150:
>
> 5060;rport=5060;branch=z9hG4bKBAF99F2231994F0D9A53836019EAC108..Record-Route:
> <sip:sip.prodec.tv:5060;maddr=80.234
>
> .135.99>..Record-Route:
> <sip:3018@161.30.94.136;ftag=399847332;lr=on>..From: "Dave Bath"
> <sip:admin@sip.dev.inmars
>
> at.com>;tag=399847332..To:
> <sip:3018@sip.prodec.tv>;tag=q5elumfa3r..Call-ID:
> FC2BDDD4-9E47-4C5A-9034-4977FC86290D@
>
> 161.30.94.150..CSeq: 35761 INVITE..Contact:
> <sip:3018@sip.prodec.tv;gruu=do1iiw55>..Allow: INVITE, ACK, CANCEL, BY
>
> E, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE,
> INFO..Content-Length: 0....
>
> -------------------- END dialling full sip address ----------------
>
> -------------------- BEGIN dialling pdt code ----------------------
>
> #
>
> U 161.30.94.150:5060 -> 161.30.94.136:5060
>
> INVITE sip:839503018@sip.dev.inmarsat.com SIP/2.0..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport;branch=z9hG4bK4702083
>
> ECD1D437DA04923E23027A6A5..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=570031081..To: <sip:839503018@sip.
>
> dev.inmarsat.com>..Contact: <sip:admin@161.30.94.150:5060>..Call-ID:
> EE63DA9A-C436-4598-B636-F3C8368E711E(a)161.30.9
>
> 4.150..CSeq: 64028 INVITE..Max-Forwards: 70..Content-Type:
> application/sdp..User-Agent: X-Lite release 1103m..Cont
>
> ent-Length: 298....v=0..o=admin 434296285 434296325 IN IP4
> 161.30.94.150..s=X-Lite..c=IN IP4 161.30.94.150..t=0 0.
>
> .m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0
> pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:9
>
> 8 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-15..
>
> #
>
> U 161.30.94.136:5060 -> 161.30.94.150:5060
>
> SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport=5060;branch=z9hG4bK4
>
> 702083ECD1D437DA04923E23027A6A5..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=570031081..To: <sip:83950301
>
> 8(a)sip.dev.inmarsat.com>..Call-ID:
> EE63DA9A-C436-4598-B636-F3C8368E711E@161.30.94.150..CSeq: 64028
> INVITE..Server:
>
> Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning:
> 392 161.30.94.136:5060 "Noisy feedback tells
>
> : pid=13743 req_src_ip=161.30.94.150 req_src_port=5060
> in_uri=sip:839503018@sip.dev.inmarsat.com out_uri=sip:3018
>
> @sip.prodec.tv. via_cnt==1"....
>
> #
>
> U 161.30.94.136:5060 -> 80.234.135.99:5060
>
> INVITE sip:3018@sip.prodec.tv. SIP/2.0..Record-Route:
> <sip:839503018@161.30.94.136;ftag=570031081;lr=on>..Via: SIP
>
> /2.0/UDP 161.30.94.136;branch=z9hG4bK705d.1d3e9f23.0..Via: SIP/2.0/UDP
> 161.30.94.150:5060;rport=5060;branch=z9hG4b
>
> K4702083ECD1D437DA04923E23027A6A5..From: Dave Bath
> <sip:admin@sip.dev.inmarsat.com>;tag=570031081..To: <sip:839503
>
> 018(a)sip.dev.inmarsat.com>..Contact:
> <sip:admin@161.30.94.150:5060>..Call-ID:
> EE63DA9A-C436-4598-B636-F3C8368E711E@
>
> 161.30.94.150..CSeq: 64028 INVITE..Max-Forwards: 69..Content-Type:
> application/sdp..User-Agent: X-Lite release 110
>
> 3m..Content-Length: 298....v=0..o=admin 434296285 434296325 IN IP4
> 161.30.94.150..s=X-Lite..c=IN IP4 161.30.94.150
>
> ..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0
> pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=
>
> rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-15..
>
> #
>
> U 80.234.135.99:5060 -> 161.30.94.136:5060
>
> SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> 161.30.94.136;branch=z9hG4bK705d.1d3e9f23.0..Via: SIP/2.0/UDP
> 161.30.94.150:5
>
> 060;rport=5060;branch=z9hG4bK4702083ECD1D437DA04923E23027A6A5..From:
> "Dave Bath" <sip:admin@sip.dev.inmarsat.com>;
>
> tag=570031081..To: <sip:839503018@sip.dev.inmarsat.com>..Call-ID:
> EE63DA9A-C436-4598-B636-F3C8368E711E(a)161.30.94.1
>
> 50..CSeq: 64028 INVITE..Content-Length: 0....
>
> #
>
> ------- END dialling with pdt module ---------------------------
>
> In the second case, there is nothing futher transmitted until a
> request timed out is received.
>
> Many thanks for looking into this...
>
> Dave
>
> -----Original Message-----
> From: Daniel-Constantin Mierla [mailto:daniel@iptel.org]
> Sent: 10 August 2004 10:10
> To: Dave Bath
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] PDT module
>
> we need network traffic dumps (ngrep, ethereal) to see what happens there.
>
> Daniel
>
> On 8/9/2004 4:23 PM, Dave Bath wrote:
>
>>
>
>>
>
>>I have been trying to use the pdt module, and the idea seems
> excellent. I have the database up and running, and can enter codes.
> Things seem to be translated correctly, but the INVITE request is not
> complete, and the call always fails. Compare the following from the
> log file:
>
>>
>
>>
>
>>
>
>>ACC: transaction answered: method=INVITE,
> i-uri=sip:839503018@sip.dev.inmarsat.com, o-uri=sip:3018@sip.prodec.tv
>
>>
>
>>
>
>>
>
>>ACC: transaction answered: method=INVITE,
> i-uri=sip:3018@sip.prodec.tv, o-uri=sip:3018@sip.prodec.tv,
> call_id=9C912086-C197-484D-8AD2-E261F1A3234A(a)161.30.94.150, from=Dave
> Bath <sip:admin@sip.dev.inmarsat.com>;tag=2122510239, code=487
>
>>
>
>>
>
>>
>
>>The first is dialing using the pdt module, with my code
> 8+395+0+<number> the second is directly dialing the full external
> domain address. The first fails, and I eventually get a timeout, the
> second connects no problem.
>
>>
>
>>
>
>>
>
>>Excerpts from the ser.cfg are:
>
>>
>
>>
>
>>
>
>> # loose-route processing
>
>>
>
>> if (loose_route()) {
>
>>
>
>> t_relay();
>
>>
>
>> break;
>
>>
>
>> };
>
>>
>
>>
>
>>
>
>> # we record-route all messages -- to make sure that
>
>>
>
>> # subsequent messages will go through our proxy; that's
>
>>
>
>> # particularly good if upstream and downstream entities
>
>>
>
>> # use different transport protocol
>
>>
>
>> record_route();
>
>>
>
>>
>
>>
>
>> # label all transaction for accounting
>
>>
>
>> setflag(1);
>
>>
>
>>
>
>>
>
>> # Make sure we check the prefix tables
>
>>
>
>> prefix2domain();
>
>>
>
>>
>
>>
>
>>before the if (uri==myself) statement.
>
>>
>
>>
>
>>
>
>>I have been bashing my head around and around for so long now, do you
> have any ideas?!
>
>>
>
>>
>
>>
>
>>Many many thanks in advance,
>
>>
>
>>
>
>>
>
>>Dave
>
>>
>
>>
>
>>
>
>> /-------------------------------------/
>
>>
>
>> /Dave Bath/
>
>>
>
>> /Inmarsat LTD/
>
>>
>
>> /Global Satellite Communications/
>
>>
>
>> /www.inmarsat.com <http://www.inmarsat.com/> /
>
>>
>
>> /Regional BGAN Engineer/
>
>>
>
>> /07736 232085/
>
>>
>
>> NOTE: The information contained in this email is intended for the
>
>> named recipients only, it may be privileged and confidential. If you
>
>> are not the intended recipient, you must not copy distribute or take
>
>> any action in reliance upon it. No warranties or assurances are made
>
>> in relation to the safety and content of this email and any
>
>> attachments. No liability is accepted for any consequences arising
> from it
>
>>
>
>>
>
>>
>
>>------------------------------------------------------------------------
>
>>
>
>>_______________________________________________
>
>>Serusers mailing list
>
>>serusers(a)lists.iptel.org
>
>>http://lists.iptel.org/mailman/listinfo/serusers
>
>>
>
>>
>
Hi all,
i just installed mysql from www.mysql.com into my linux for supporting
my ser database. i don't know how to point this mysql to ser system
i stuck now!! pls help me how to make it!!
thank in advance any help
Br/Zahari
AtlasONE Sdn Bhd
Greetings All,
I am having a small problem that I'm not sure what the cause is. I'm running
ser, serweb, and sems. After loading the voicemail module in the ser.cfg, I
am receiving "Unable to connect to database, too many connections." MySQL
tells me there are 101 threads running. This only happens when I load the
vm.so modules. I have successfully been able to get it to load, but now
things like serweb will not work with the same error.
I have missed something and/or set the configuration of sems or ser wrong?
Or do I need just up the number of connections limit to mySQL? (which I have
never done, any pointers?)
Thanks and have fun.
C.
what is asterisk??
could somebody give me some information to help me!
Thanks.
>SEMS could not do this currently, you would need some IVR based
>voicemail system I guess, for example asterisk.
>Jan.
>>On 06-08 12:06, Sean wrote:
>> When I record voice and save wav file, it will send mail to my user.
>> But I don't want to do that.I want it save in my server and when user
>> register
>> my server,my server will send message to tell him.He have a miss call.How
to
>> do it?
>> Could anybody help me? Thanks.
>>
>> --
>> National Central Univ., Computer Center (http://www.cc.ncu.edu.tw/index
php)
>>
>> _______________________________________________
>> Serusers mailing list
>> Serusers at iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
Hi all,
My SER is configured to work with authenticatoin, something like this:
if (method=="INVITE") {
proxy_authorized(".....
But in case of re-INVITE during the same session SER is requiring authenticatoin again.
So, my question is how to avoid authentication request from SER in case of re-INVITE.
Thanks in advance
Alex Zontik
hi,all,
I want to use other equipment to strengthen ser's function. And it is needed to add voicexml script into the exiting sip messages. But I don't know how to do.
Who is so appreciated to tell me? Thank you!
Hi all,
My SER is configured to work with authenticatoin, something like this:
if (method=="INVITE") {
proxy_authorized(".....
But in case of re-INVITE during the same session SER is requiring authenticatoin again.
So, my question is how to avoid authentication request from SER in case of re-INVITE.
Thanks in advance
Alex Zontik
I know SER is a pure Proxy server instead of a B2BUA, I know iptel has commercial Application Agent (AA) module to take care pre-paid scenario, but beside that, is there have any alternative way to tear down a call (just drop a call - the most basic pre-paid request, don't need forward to IVR etc. other complex process) from SER system by any configuration, external script or source code hacking?
Some of approach I can imagine are:
1/ Trace each call session with external script, and then issue properly BYE message to both caller and callee party when need shutdown a call; but I don't know how to take care the call sesion with external program access, and how to generate BYE message;
2/ If full RTP Proxy mode, we can shutdown the rtp proxy path for that call session to stop the voice conversation, in this way caller or callee will hangup for no voice stream problem, it could be an alternative way to tear down a call; but same question how to monitor that call and then issue command to RTPPROXY for disconnecting the voice path?
Hope my questions and suggestions make sense to you, thank you for you patient and kindly reply.
Regards,
--
_______________________________________________
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most clients just register ONE server, so can SER register with SEVERAL and
forward to the G/W's as needed. As SER cannot easily generate messages , then
the REGISTER message must somehow be forwarded..and applied to itself and others.
no mention of this OBVIOUS need anywhere that i have looked yet.
thx
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hello friends ,
iam using version: ser 0.8.12-1rc6 (i386/linux)
,cpl-c module and CPLEd to upload to the
server and the databse.
it s been storing in database of ser and in cpl table
7(2921) DEBUG:parse_accept_hdr: missing Accept header
7(2921) XLOG: xl_print_log: final buffer length 29
7(2921) Error: fail cpl registering
7(2921) receive_msg: cleaning up
so it is not registering the cpl script
can i get any leads in this
with regards
serdiehard
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