hello friends,
does we have any free windows based softphones with
g723 codec which run with sip express router
with regards
rama kanth
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(Sorry if this is reposted, but noticed I had some failure reports for
this, so not sure if it made it to the mailing list or not)
Hey there Andrei (or anyone else!),
Here are the syslog messages and ngrep for a serweb send message.
Serweb reports message sent ok, but I don't receive it on the client,
nor is it stored in the Message Store?
Any ideas would be very welcome!
Dave
MSILO: the downstream UA does not support MESSAGE requests ...
Jul 20 08:49:49 sip /usr/sbin/ser[23058]: ACC: transaction answered:
method=MESSAGE, i-uri=sip:admin@161.30.94.68,
o-uri=sip:admin@161.30.94.70:5060, call_id=73f9fea9-0(a)127.0.0.1,
from=sip:test1@sip.dev.inmarsat.com;tag=533cb9e91f4b999cf76861cbb9ed54ed
-fba5, code=202
Jul 20 08:49:52 sip /usr/sbin/ser[23017]: ERROR: udp_send:
sendto(sock,0xbd777de0,667,0,0xbd779394,16): Invalid argument(22)
Jul 20 08:49:52 sip /usr/sbin/ser[23017]: CRITICAL: invalid
sendtoparameters one possible reason is the server is bound to localhost
and attempts to send to the net
Jul 20 08:49:52 sip /usr/sbin/ser[23017]: msg_send: ERROR: udp_send
failed
Jul 20 08:49:52 sip /usr/sbin/ser[23017]: ERROR: t_forward_nonack:
sending request failed
Jul 20 08:49:52 sip /usr/sbin/ser[23017]: ACC: transaction answered:
method=MESSAGE, i-uri=sip:daemon@mydomain.org,
o-uri=sip:daemon@mydomain.org, call_id=73f9fea8-0(a)161.30.94.68,
from=sip:sip_registrar@161.30.94.68;tag=533cb9e91f4b999cf76861cbb9ed54ed
-1247, code=477
ngrep UDP port 5060
interface: eth0 (161.30.94.64/255.255.255.224)
filter: ip and ( port 5060 )
match: UDP
###########################
U 161.30.94.68:5060 -> 161.30.94.70:5060
MESSAGE sip:admin@161.30.94.70:5060 SIP/2.0..Max-Forwards:
10..Record-Route
:
<sip:admin@161.30.94.68;ftag=533cb9e91f4b999cf76861cbb9ed54ed-fba5;lr=on
>
..Via: SIP/2.0/UDP 161.30.94.68;branch=z9hG4bK33b2.acca75b7.0..Via:
SIP/2.0
/UDP 161.30.94.68;branch=z9hG4bKb789.54d3c0e.0..To:
<sip:admin@161.30.94.68
>..From:
sip:test1@sip.dev.inmarsat.com;tag=533cb9e91f4b999cf76861cbb9ed54e
d-fba5..CSeq: 10 MESSAGE..Call-ID: 73f9fea9-0@127.0.0.1..Content-Length:
16
..User-Agent: Sip EXpress router(0.8.12 (i386/linux))..p-version:
Web_inter
face_Karel_Kozlik-0.9..Contact: <sip:daemon@mydomain.org>..Content-Type:
te
xt/plain; charset=UTF-8....test message....
##
U 161.30.94.70:5060 -> 161.30.94.68:5060
SIP/2.0 405 Method Not Allowed..Via: SIP/2.0/UDP
161.30.94.68;branch=z9hG4b
K33b2.acca75b7.0..Via: SIP/2.0/UDP
161.30.94.68;branch=z9hG4bKb789.54d3c0e.
0..From:
sip:test1@sip.dev.inmarsat.com;tag=533cb9e91f4b999cf76861cbb9ed54e
d-fba5..To: <sip:admin@161.30.94.68>;tag=3946696931..Contact:
<sip:admin@16
.30.94.70:5060>..Call-ID: 73f9fea9-0@127.0.0.1..Allow:
INVITE,ACK,BYE,CANC
EL,OPTIONS,NOTIFY..CSeq: 10 MESSAGE..Server: X-Lite release
103m..Content-
Length: 0....
#exit
31 received, 0 dropped
-----Original Message-----
From: Andrei Pelinescu-Onciul
[mailto:pelinescu-onciul@fokus.fraunhofer.de]
Sent: 19 July 2004 15:37
To: Dave Bath
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Problem with messages + msilo
On Jul 19, 2004 at 13:32, Dave Bath <dave(a)fuuz.com> wrote:
> Hey there,
>
> Once again, many thanks for a swift response Andrei! No, the machine
> does not have two interfaces, it's a single public ip address. What do
> you mean about ingress filtering? Which network dumps would be
helpful?
If it's a single ip, than don't worry about ingress filtering.
Just dump the udp traffic to your machine port 5060 until you get one of
the log error message (sendto ...) and then send use the log and the
dump.
The message might also be caused by some UA using a 0 source port (which
is highly broken), or a 0 destination port or ip in the message uri.
Andrei
Hi,
When i tried to run rtpproxy as non root, it gave an
error "rtpproxy: can't bind to a socket: Permission
denied". Is there a reason to run it as root?
Thanks,
Richard
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Hi all,
I need to configure two ser servers to do such a thing :
SER2 is the registrar proxy, knowing ips ...
SER1 is a must go through proxy. Each packet must go through him before being
sent to SER2.
________ ________
| | REGISTER | |
Phone B ---->| |------------------>| |
| SER1 | | SER2 |
| | | |
| IP .4 | REGISTER | IP .8 |
Phone A ---->| |------------------>| |
|_______ | |________|
CALL from A to B
________ ________
| | CALL | |
Phone B <----| |<------------------| |
| SER1 | | SER2 |
| | | |
| IP .4 | CALL | IP .8 |
Phone A ---->| |------------------>| |
|_______ | |________|
When a call is done by A, it must go through ser1 then to ser2 and before
coming back to phone B it must go to SER1.
I can't find a way to do that with a script. If someone can help me, it would
be great.
Thanks,
Laurent
I'm building a system that "sits" both in the LAN and in the Wan.
My question is this:
Is there a way to check the IP address of the person I'm trying to
contact?
So when a LAN user contacts a LAN user or WAN (not behind Nat) user
contacts a WAN (not behind Nat) user
They will be no use of the RTP proxy?
Something like: if (contact_ip_addr != src_ip ) { ......};
Thanx
Shi Hoch
There are 3 possible points where the private IP might have been replaced with the ADSL gateway's public address.
1. On the client itself (using STUN?)
2. On the ADSL gateway (Is it SIP aware?)
3. Misconfigured SER (I think this is not the case - since it works for port 5070).
To confirm if it is (1) - run a sniffer like ethereal or tcpdump to capture packets as it leaves the client machine - both for port 5060 and 5070. See if the client is doing anything smart - replacing private IP with gateway IP while using 5060 and not for 5070?
To confirm if it is (2) - run a sniffer on the same collission domain or on the same machine where SER is running and capture the registration request packets. If (1) is false and still the private IP is getting modified, its the ADSL gateway that's doing some proxying of SIP traffic (port 5060).
Which ADSL gateway are you using?
Dhiraj
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On
Behalf Of Bart Van Daal
Sent: 20 July 2004 09:16
To: serusers(a)lists.iptel.org
Subject: RE: [Serusers] NAT vs. NoNat authentication
Hello I'll post the answers to the two replies:
>Andrei> Do you have another UA behind the same nat, using 5060?
No it's only 1 Phone ---- ADSLRouter(NAT) ------- Internet ---------- Ser
Dhiraj > ..i'l post the two ngreps again:
I'm sorry for the long post. What I can see is, when ser runs on port 5070
the register contains a private ip in the 'Via:' header. When it runs on
5060
The 'Via:' header contains the public IP of the router and an unprivileged
port.
----------------------------------------------- 5070
-----------------------------------------------
filter: ip and ( port 5070 )
#
U 213.219.137.137:5070 -> 212.71.0.60:5070
REGISTER sip:ser.edpnet.net:5070 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.2:5070.
Supported: replaces.
User-Agent: SIP201 (lp201sip.100a).
Contact: <sip:bart@10.0.0.2:5070>;expires=60.
From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-0-42e-7fea.
To: <sip:bart@ser.edpnet.net>.
Call-ID: a000002-13ce-0-406-79bf-1.
CSeq: 1 REGISTER.
Content-Length:0.
.
#
U 212.71.0.60:5070 -> 213.219.137.137:5070
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 10.0.0.2:5070;rport=5070;received=213.219.137.137.
From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-0-42e-7fea.
To: <sip:bart@ser.edpnet.net>;tag=2497a39c629b119dac83769f58cd2b29.1cd2.
Call-ID: a000002-13ce-0-406-79bf-1.
CSeq: 1 REGISTER.
WWW-Authenticate: Digest realm="ser.edpnet.net",
nonce="40fcce4ec4ab3796c95cb2c87a9d94a05651ed08".
Server: Sip EXpress router (0.8.13-dev-33-usrloc (i386/linux)).
Content-Length: 0.
Warning: 392 212.71.0.60:5070 "Noisy feedback tells: pid=18743
req_src_ip=213.219.137.137 req_src_port=5070 in_uri=sip:ser.edpnet.net:5070
out_uri=sip:ser.edpnet.net:5070 via_cnt==1".
.
#
U 213.219.137.137:5070 -> 212.71.0.60:5070
REGISTER sip:ser.edpnet.net:5070 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.2:5070.
Supported: replaces.
User-Agent: SIP201 (lp201sip.100a).
Contact: <sip:bart@10.0.0.2:5070>;expires=60.
Authorization: Digest username="bart", realm="ser.edpnet.net",
nonce="40fcce4ec4ab3796c95cb2c87a9d94a05651ed08",
uri="sip:ser.edpnet.net:5070", response="ea0329c8f3a4d199230733feb750d3a1",
algorithm=MD5.
From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-40fccd8d-1991-7051.
To: <sip:bart@ser.edpnet.net>.
Call-ID: a000002-13ce-0-406-79bf-1.
CSeq: 2 REGISTER.
Content-Length:0.
.
#
U 212.71.0.60:5070 -> 213.219.137.137:5070
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.0.0.2:5070;rport=5070;received=213.219.137.137.
From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-40fccd8d-1991-7051.
To: <sip:bart@ser.edpnet.net>;tag=2497a39c629b119dac83769f58cd2b29.1cd2.
Call-ID: a000002-13ce-0-406-79bf-1.
CSeq: 2 REGISTER.
Contact: <sip:bart@213.219.137.137:5070>;expires=60.
Server: Sip EXpress router (0.8.13-dev-33-usrloc (i386/linux)).
Content-Length: 0.
Warning: 392 212.71.0.60:5070 "Noisy feedback tells: pid=18743
req_src_ip=213.219.137.137 req_src_port=5070 in_uri=sip:ser.edpnet.net:5070
out_uri=sip:ser.edpnet.net:5070 via_cnt==1".
.
----------------------------------------------- 5060
-----------------------------------------------
filter: ip and ( port 5060 )
#
U 213.219.137.137:5060 -> 212.71.0.60:5060
REGISTER sip:ser.edpnet.net:5060 SIP/2.0.
Via: SIP/2.0/UDP 213.219.137.137:47726.
Supported: replaces.
User-Agent: SIP201 (lp201sip.100a).
Contact: <sip:bart@10.0.0.2:5060>;expires=60.
From: <sip:bart@ser.edpnet.net> ;tag=a000002-13c4-0-429-495.
To: <sip:bart@ser.edpnet.net>.
Call-ID: a000002-13c4-0-401-719e-1.
CSeq: 1 REGISTER.
Content-Length:0.
.
#
U 212.71.0.60:5060 -> 213.219.137.137:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 213.219.137.137:47726;rport=5060.
From: <sip:bart@ser.edpnet.net> ;tag=a000002-13c4-0-429-495.
To: <sip:bart@ser.edpnet.net>;tag=61a88e7fd5f0561d96cde0cc9ecba6d7.2508.
Call-ID: a000002-13c4-0-401-719e-1.
CSeq: 1 REGISTER.
WWW-Authenticate: Digest realm="ser.edpnet.net",
nonce="40fcccbe3b4e06bc429de0a886d7b43409cb8427".
Server: Sip EXpress router (0.8.13-dev-33-usrloc (i386/linux)).
Content-Length: 0.
Warning: 392 212.71.0.60:5060 "Noisy feedback tells: pid=18727
req_src_ip=213.219.137.137 req_src_port=5060 in_uri=sip:ser.edpnet.net:5060
out_uri=sip:ser.edpnet.net:5060 via_cnt==1".
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Ser has a module cpl-c which supports cpl.
There is a rfc for cpl, just google it. It's very easy
to read.
--- Joseph Cheung <Joseph.Cheung2(a)TELUS.COM> wrote:
> Question now is how to get the support for cpl?
>
> Can you point me to the right uri?
>
> Thanks
> -----Original Message-----
> From: Richard [mailto:mypop3mail@yahoo.com]
> Sent: July 19, 2004 9:51 PM
> To: Joseph Cheung; serusers(a)lists.iptel.org
> Subject: Re: [Serusers] developer related question
>
> You can use a cpl to each subscriber. Then it can
> forward, redirect or reject the request according to
> the cpl script, which is a much flexible solution.
>
> Richard
>
>
> --- Joseph Cheung <Joseph.Cheung2(a)TELUS.COM> wrote:
> > Hi.
> >
> > I think there is another email address I would use
> > for this kinds of question. If so, much appreciate
> > if someone could let me know.
> >
> > Here is my question.
> >
> > I have created a module that will lookup the
> > database and determine if I should forward,
> > redirect, or reject the request.
> >
> > Since there is no switch statement in the script,
> I
> > am wondering if I would save the "action" into a
> > static variable and export a function to check and
> > determine the action.
> >
> > I know the above could be done and it should be ok
> > as we do a fork - should duplicate the memory as
> > well.
> >
> > I would much appreciate if someone could give me a
> > YES or NO answer or any idea how this is done.
> >
> > Many thanks.
> >
> > -----Original Message-----
> > From: serusers-bounces(a)lists.iptel.org
> > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of
> > Jiri Kuthan
> > Sent: June 19, 2004 2:01 PM
> > To: Steve Blair; serusers(a)lists.iptel.org
> > Subject: Re: [Serusers] CC-Diversion
> >
> > This may help:
> >
>
http://www.iptel.org/ser/doc/seruser/seruser.html#AEN1018
> >
> > -jiri
> >
> > At 04:09 PM 6/17/2004, Steve Blair wrote:
> >
> > > Hello:
> > >
> > > Can anyone speak to how the CC-Diversion
> field
> > is used
> > >by Cisco gateways? It is my understanding that
> > adding this header
> > >field in SER will result in the Cisco gateway
> > setting the calling party
> > >id to the value in the CC-Diversion header. Is
> this
> > correct?
> > >
> > > I'm asking because we have an Octel 350 voice
> > mail system that
> > >I would like to use for mailbox for IP phone
> users.
> > The 350 expects
> > >a physical SMDI circuit for the call signaling
> path
> > and a different
> > >circuit for the message body path. It would be
> > great to allow the
> > >SIP signaling exiting our IP cloud through a
> Cisco
> > gateway to be
> > >able to tell the Octel system to which subscriber
> > mailbox the message
> > >should be delivered.
> > >
> > >Thanks,Steve
> > >
> > >--
> > >
> > >ISC Network Engineering
> > >The University of Pennsylvania
> > >3401 Walnut Street, Suite 221A
> > >Philadelphia, PA 19104
> > >
> > >
> > >voice: 215-573-8396
> > > 215-746-7903
> > >
> > >fax: 215-898-9348
> > >
> > >sip:blairs@upenn.edu
> > >
> > >_______________________________________________
> > >Serusers mailing list
> > >serusers(a)lists.iptel.org
> > >http://lists.iptel.org/mailman/listinfo/serusers
> >
> > --
> > Jiri Kuthan http://iptel.org/~jiri/
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
>
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Only SIP messages. Signalling and RTP have completely independent
paths.
Unclassified
>>> "Lakmal Silva" <lakmal(a)lankacom.net> 07/20/04 06:27AM >>>
Hi all,
I have a question on record routing. Once record routing is done, will
the
whole traffic (including rtp) will be routed through SER or only all
the
SIP messages will be passed through SER?
--
Regards,
Lakmal
Lankacom Services (Pvt) Ltd.
65C, Dharmapala Mawatha,
Colombo 07.
Sri Lanka.
Tel: +94-11-2437545
www.lankacom.net
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi,
I've got ser working great over here, but I cannot find out how I can
play a message before forwarding the call. Like this:
- Person A calls person B via the SER proxy
- SER proxy forwards to mediaserver
- Mediaserver plays a message (like 'this call is free of cost')
- Mediaserver hangs up
- SER proxy forwards the call to person B
- Person A and person B can talk to each other now..
Is this possible ?
Thanks,
Leon de Rooij
leon(a)scarlet-internet.nl
Thanks you all of you who replied so promptly.
> Only SIP messages. Signalling and RTP have completely independent
> paths.
>
> Unclassified
>>>> "Lakmal Silva" <lakmal(a)lankacom.net> 07/20/04 06:27AM >>>
> Hi all,
>
> I have a question on record routing. Once record routing is done, will
> the
> whole traffic (including rtp) will be routed through SER or only all
> the
> SIP messages will be passed through SER?
>
>
> --
> Regards,
> Lakmal
>
> Lankacom Services (Pvt) Ltd.
> 65C, Dharmapala Mawatha,
> Colombo 07.
> Sri Lanka.
> Tel: +94-11-2437545
> www.lankacom.net
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
--
Regards,
Lakmal
Lankacom Services (Pvt) Ltd.
65C, Dharmapala Mawatha,
Colombo 07.
Sri Lanka.
Tel: +94-11-2437545
www.lankacom.net
