siproxd-users — Users Mailing List

[Siproxd-users] gethostbyname error

[Simon M. <sim...@ma...>]

I just installed siproxd-0.5.11 on my Gentoo Linux machine.

I tried to make an outgoing call, but I get the following error:

Aug  1 05:28:01 firewall siproxd: register.c:71 WARNING:registration file 
not found, starting with empty table
Aug  1 05:28:01 firewall siproxd: siproxd.c:262 INFO:siproxd-0.5.11-4 
i586-pc-linux-gnu started
Aug  2 23:28:08 firewall siproxd: utils.c:184 
ERROR:gethostbyname(fwd.pulver.com) failed: h_errno=0 [Resolver Error 0 
(no error)]
Aug  2 23:28:10 firewall siproxd: proxy.c:211 INFO:Outgoing Call from: 
89884@192.168.89.13
Aug  2 23:28:14 firewall siproxd: proxy.c:211 INFO:Outgoing Call from: 
89884@192.168.89.13
Aug  2 23:28:25 firewall siproxd: utils.c:184 
ERROR:gethostbyname(fwd.pulver.com) failed: h_errno=0 [Resolver Error 0 
(no error)]


On the same machine, I can ping the host it is complaining about:
# ping fwd.pulver.com
PING fwd.pulver.com (69.90.155.70) 56(84) bytes of data.
64 bytes from 69.90.155.70: icmp_seq=1 ttl=52 time=108 ms

Any ideas?

Simon

Re: [Siproxd-users] zyxel 2000W config

[Peter K. <pe...@as...>]

On Sat, 21 May 2005, Andreas Jellinghaus wrote:

> Peter, could you please post your config file
> (and maybe also the settings you did on your zyxel 2000w)?
>
> I'd like to use that phone, too, with sipproxyd and sipgate,
> I too have a qsc dsl net connection, so your config should
> help me a lot.
>
> I will recompile siproxyd and remove the lines suggested
> to workaround the sipgate issue.


hi!

After some more experience I strongly suggest not to use the Zyxel
P2000W. It works somehow, but not reliably so. At present, I can call
in from my landline, but don't even reach the 10000 test number from
here.
I did an update to the latest zyxel firmware, but it is still not quite the
best firmware I ever saw.

Really, don't waste your money on this.

Hope that helps...


-- 
peter koellner <pe...@as...>

[Siproxd-users] zyxel 2000W config

[Andreas J. <aj...@du...>]

Peter, could you please post your config file
(and maybe also the settings you did on your zyxel 2000w)?

I'd like to use that phone, too, with sipproxyd and sipgate,
I too have a qsc dsl net connection, so your config should
help me a lot.

I will recompile siproxyd and remove the lines suggested
to workaround the sipgate issue.

Thanks for your help.

Regards, Andreas
p.s. sorry for spamming the list, but the sf.net webarchive
of this list hides his email address, so I can't contact him
directly.

[Siproxd-users] Problem with sound!

[serge m. <ser...@ya...>]

Hi all


       I use siproxd as outbound proxy and in front of
a router . Signalisation work well but i have a sound
problem. When the callee accept the call, there is no
sound in each client. 

 here is the configuration of my siproxd.conf

if_inbound  = eth0
if_outbound = eth0
host_outbound = 62.128.173.99
hosts_allow_reg = 192.168.0.0/24
sip_listen_port = 5075
daemonize = 1
silence_log = 1
log_calls = 1
user = siproxd
registration_file =
/var/lib/siproxd/siproxd_registrations
autosave_registrations = 300
rtp_proxy_enable = 1
rtp_port_low  = 7070
rtprtp_timeout = 300
_port_high = 7079
rtp_dscp = 46
default_expires = 600
debug_level = 0
debug_port = 0

     Regards

                         Serge MESSA







	
	
		
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[Siproxd-users] Help for determining Next-Hop Address

[harry g. <gai...@ya...>]

Hi all,

I need help to force siproxd to fetch address/port
from topmost Route when message 100 180 must be sent.

May be the problem is in proxy.c file.

I need help to fix this problem .

Regards
Harry


	

	
		
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[Siproxd-users] Problem with siproxd!

[serge m. <ser...@ya...>]

Hi all, 

1-    I  use siproxd as a out bound proxy. The
installation passed well ,but it not work for me! I
use the xlite client and i make registrations with the
vocal-1.5.0 of vovida. I'm not find the file
siproxd.pid in any location in my pc.I use linux
fedora core 2.


2-   I want to know really how the sip packets walks
on the network from the first client to the other one
through the 3rd registration party and the out bound
proxy (siproxd) when a call is made!

   Please, help me! 

  Thanks in advance

                   Serge MESSA


	

	
		
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[Siproxd-users] I use siproxd-0.5.11 as out bound proxy and it not work for me!

[serge m. <ser...@ya...>]

Hi all!

    I install successfully the siproxd-0.5.11 and i
want to use it as a outbound router on my linux fedora
core 2 PC ,but it not work for me. I cannot make a
call! 
may be a configuration problem in the
/usr/local/etc/siproxd.conf or a xlite configuration.

1-the lines below are enabled in the siproxd.conf file

if_inbound  = eth0
if_outbound = eth0
host_outbound = 62.128.173.99
sip_listen_port = 5060
daemonize = 1
silence_log = 1
log_calls = 1
user = nobody  
registration_file =
/var/lib/siproxd/siproxd_registrations
autosave_registrations = 300
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 8000
rtp_port_high = 8000
rtp_timeout = 300
rtp_dscp = 46
default_expires = 600
debug_level =      0x00000000
debug_port = 0

 i use the xlite client . 

2-xlite client configuration
in menu->sip proxy->default
Domain/Realm: <host part of SIP URI>
SIP Proxy: <IP address of 3rd party registrar>
Out Bound Proxy: <inbound IP address of siproxd>
Use Out Bound Proxy: Always

in menu->network
NAT firewall IP: <Public ip adress of my  router>
Force Firewall Type: Open IP

Help me please;

             Thanks in advance

                                serge MESSA



	

	
		
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[Siproxd-users] i want to subscribe siproxd-users mail list.

[caixu <ca...@sj...>]

IA==

[Siproxd-users] siproxd NOT on NAT/Firewall

[krishnamurthy k. <mur...@ya...>]

Hi,  
i would like to install siproxd on a computer behind  
my firewall/router, is this scenario working with current release of siproxd. can any one help me how to make work this scenario using siproxd .Thank you in advance.  
 
Regards,  
kkmurthy  




		
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[Siproxd-users] running siproxd "in front" of a NAT router

[Thomas R. <tr...@gm...>]

Hi there,

A number of people have asked if it is not possible to run siproxd
on a host *inside* the private network and *not* of the NAT router
itself. The reason for these requests is, most of home users use some
out-of-the-box ADSL router that can not run any additional software.

I'm currently working to implement such functionality into siproxd,
so it can be run "in front" (or behind - depends on the point of view)
of an NAT router:

  local (private)
  network
                       +-------------+
-----+------------+----| ADSL router |------------>> Internet
     |            |    +-------------+
     |            |
  +------+    +-------+
  | SIP  |    |siproxd|
  |client|    +-------+
  +------+


Anyone looking for this feature is encouraged to download the latest
snapshot. Currently no documentation exists, but following I include
a short list of things to consider.


On your Router:
- forward incoming data for UDP/5060 to the siproxd host
- forward incoming data for UDP/7070-7079 to the siproxd host

Siproxd config:
- set both, 'if_inbound' and 'if_outbound' to the inbound interface
  (usually the only network interface on that box)
- additionally define 'host_outbound' with the IP address (or 
  hostname) of your public IP. You may want use a DYNDNS hostname
  to be able to deal with dynamically changing IP addresses.

I was able to place an outgoing call via sipphone.com to my normal
telephone line and got two way audio - it can't be that bad then :-)

Something that currently will fail is using symmetric RTP. If you can
configure this in your SIP client, disable it.

/Thomas


-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:174...@pr... | sip:43...@fw...

[Siproxd-users] siproxdy capabilities/configuration

[Ant <an...@sy...>]

Hi,

Im currently in the process of replacing the masquerading firewall at my
work. Currently this firewall has ser installed on it, and internal
office calls get routed through it to our internal pbx host. 

However, we wish to be able to make incoming and outgoing sip calls
between work and our homes. The configuration looks like this:
(hopefully the ascii comes out). The hardphones are Polycom SIP phones.

                                       ISDN
                                        |
      +---------------------+    +------+-----+   +----------+
Net---|masq firewall/siproxd|----|Software PBX|-+-|Hardphone1|
      +---------------------+    +------------+ | +----------+
                                                |
                                                | +----------+
                                                +-|Hardphone2|
                                                  +----------+

Looking at siproxd docs, it seems that normally we would configure our
hardphones to register with siproxd for incoming calls via the net. In
our case though our PBX provides functionality such as voicemail, as
well as ISDN connectivity. Is it possible to configure siproxd to proxy
any incoming sip calls to our domain through our PBX, as well as
proxying outbound sip calls over the net? 

In this scenario the phones would register with the PBX, and we hope to
be able to get sip via the net, ISDN, and also aditional PBX
functionality. Any pointers about how to set this up would be
appreciated.

Thanks,
Anthony

[Siproxd-users] Bugfix: Re-Invites and missing audio

[Thomas R. <tr...@gm...>]

All of you having problems with re-Invites and missing audio
should try the lates snapshot (uploaded a few minutes ago).

Siproxd did not properly handle changing IP addresses for the RTP
stream. I was now able to perform a SIP call to my POTS phone via
Sipphone.com - with 2-way audio working.

Feedback is welcome ;-)

/Thomas

-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:174...@pr... | sip:43...@fw...

Antwort: Re: [Siproxd-users] sipgate problem workaround

[<Mat...@mw...>]

I have tried call out through sipgate again. It also works for me now. 
Sometimes wonder happen.

Mathias Wohlfarth EDV-Beratung
Thomas-Mann-Str.1
53111 Bonn
Tel.    0172 / 53 45 591
        01801 / 777 555 33 01
Fax     0228 / 9469181
Email   mat...@mw...




Peter Koellner <pe...@as...>
Gesendet von: sip...@li...
17.01.2005 20:39
Bitte antworten an siproxd-users
 
        An:     sip...@li...
        Kopie: 
        Thema:  Re: [Siproxd-users] sipgate problem workaround


On Mon, 17 Jan 2005, Thomas Ries wrote:

> I can just try to guess what code lines you are talking about (a diff
> might have been better). IF (as I assume) the code you commented out
> lies in the function msg_make_template_reply() this would make me
> wonder, BECAUSE this function is solely used for locally processed
> requests (phone -> siproxd) and not requests directed to the provider
> (phone -> siproxd -> sipgate). A request beeing processed there will
> never be sent to the outside world.

strange... i wonder if they actually changed their authentication scheme
while i tried out changes on the code...

argghhh... now it works with the original source, so they might have 
actually changed something!!!!!

> So I would be very interested in details what you found out about
> sipgate and using VIA headers for authentication. Once the problem is
> fully understood, it should be possible to provide a
> "clean" fix/workaround that does not break other things.

well, basically the sipgate support told me that they used the VIA header
for authentication purposes and that this would not work because
siproxd added another VIA header. so I wrote them back that they should
simply check all via headers in the queue instead of the first one, but 
did not get an answer. So I guess they changed something today, and now 
it works with the regular siproxd, and it was pure coincidence that I was
experimenting with changing the source code today...

-- 
peter koellner <pe...@as...>


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Re: [Siproxd-users] sipgate problem workaround

[Peter K. <pe...@as...>]

On Mon, 17 Jan 2005, Thomas Ries wrote:

> I can just try to guess what code lines you are talking about (a diff
> might have been better). IF (as I assume) the code you commented out
> lies in the function msg_make_template_reply() this would make me
> wonder, BECAUSE this function is solely used for locally processed
> requests (phone -> siproxd) and not requests directed to the provider
> (phone -> siproxd -> sipgate). A request beeing processed there will
> never be sent to the outside world.

strange... i wonder if they actually changed their authentication scheme
while i tried out changes on the code...

argghhh... now it works with the original source, so they might have 
actually changed something!!!!!

> So I would be very interested in details what you found out about
> sipgate and using VIA headers for authentication. Once the problem is
> fully understood, it should be possible to provide a
> "clean" fix/workaround that does not break other things.

well, basically the sipgate support told me that they used the VIA header
for authentication purposes and that this would not work because
siproxd added another VIA header. so I wrote them back that they should
simply check all via headers in the queue instead of the first one, but 
did not get an answer. So I guess they changed something today, and now 
it works with the regular siproxd, and it was pure coincidence that I was
experimenting with changing the source code today...

-- 
peter koellner <pe...@as...>

Re: [Siproxd-users] sipgate problem workaround

[Thomas R. <tr...@gm...>]

Hi Peter,

I can just try to guess what code lines you are talking about (a diff
might have been better). IF (as I assume) the code you commented out
lies in the function msg_make_template_reply() this would make me
wonder, BECAUSE this function is solely used for locally processed
requests (phone -> siproxd) and not requests directed to the provider
(phone -> siproxd -> sipgate). A request beeing processed there will
never be sent to the outside world.

So I would be very interested in details what you found out about
sipgate and using VIA headers for authentication. Once the problem is
fully understood, it should be possible to provide a
"clean" fix/workaround that does not break other things.

Regards,
/Thomas


On 17 Jan, Peter Koellner wrote:
> hi!
> 
> I finally managed to make siproxd work with the zyxel P 2000W, a iptables
> firewall and sipgate.de as sip server. They really use the VIA headers
> for authentication purposes and seem to be to dumb to match against all
> entries. furthermore, they seem to be unwilling to change that.
> 
> I guess it is not totally RFC-compliant, but then the whole protocol
> structure does not seem that thoroughly thought through anyway.
> 
> I simply commented out lines 89-100 (copying VIA-headers) from
> sip_utils.c. I would like to put those lines in a SIPGATE_INSANITY
> #ifdef, or what is the preferred method to add such "specials" to the
> code?
> 
> 

-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:174...@pr... | sip:43...@fw...

[Siproxd-users] sipgate problem workaround

[Peter K. <pe...@as...>]

hi!

I finally managed to make siproxd work with the zyxel P 2000W, a iptables
firewall and sipgate.de as sip server. They really use the VIA headers
for authentication purposes and seem to be to dumb to match against all
entries. furthermore, they seem to be unwilling to change that.

I guess it is not totally RFC-compliant, but then the whole protocol
structure does not seem that thoroughly thought through anyway.

I simply commented out lines 89-100 (copying VIA-headers) from
sip_utils.c. I would like to put those lines in a SIPGATE_INSANITY
#ifdef, or what is the preferred method to add such "specials" to the
code?


-- 
peter koellner <pe...@as...>

Re: Antwort: [Siproxd-users] siproxd zyxel2000W problems

[Peter K. <pe...@as...>]

I have just tcpdumped a complete inbound call and discovered that
the local phone in fact does not send out a single RTP packet... no wonder
I don't hear anything. very strange... so I guess there must be something
wrong with the RTP information that gets to the phone on initiating the
call when picking up the phone. The sipproxy does open both rtpproxy
streams, so there must be something wrong with the message itself...

this is the OK message sent from the proxy to the phone, with 9999999 the
sip phone number, 493012345678 the land-line phone calling, 
and 192.158.99.99 the external ip number of the local net:
---------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK15a2.924f6011.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK15a2.f942f8.0
Via: SIP/2.0/UDP 195.226.161.164:5060;branch=z9hG4bK3a883ae7
Record-Route: <sip:9999999@217.10.79.9;ftag=as7fff67b2;lr=on>
Record-Route: <sip:4930869999999@217.10.79.8;ftag=as7fff67b2;lr=on>
From: "03012345678" <sip:03012345678@195.226.161.164>;tag=as7fff67b2
To: <sip:493...@si...>;tag=4D1DA4377F1766B19EF
Call-ID: 24292372190406cf552636d14167163d@195.226.161.164
CSeq: 102 INVITE
Contact: <sip:4930869999999@192.168.1.125:5060;transport=udp>
user-agent: ZyXEL P2000W VoIP Wi-Fi Phone
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, SUBSCRIBE, NOTIFY, INFO, REFER
Content-Type: application/sdp
Content-Length:   151

v=0
o=TelogyUnknown0000 59871 59871 IN IP4 195.158.99.99
s=RTP Audio
c=IN IP4 195.158.99.99
t=0 0
m=audio 7070 RTP/AVP 8
a=rtpmap:8 PCMA/8000
---------------------------------------------------------------------------



tcpdump shows me LOTS of UDP traffic on 7070 from the proxy to the phone, 
but absolutely nothing from the phone to the proxy. When I call
10000, I get bidirectional traffic...

I am not too familiar with SDP - Any wild guesses about this one? The only 
item I would find questionable would be the RTP format type. When calling
sipgates 10000, I get:

m=audio 16118 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=silenceSupp:off - - - -

sent from sipgate... and when I look at the phones voice codec list,
I see:

G.729 8k
G.711u 64k
G.711a 64k

Does this mean that the phone is not able to provide the right 
encoding for the former request since it got too few choices offered, so 
decides not to send anything? Would be weird, so perhaps not...


-- 
peter koellner <pe...@as...>

Re: Antwort: [Siproxd-users] siproxd zyxel2000W problems

[Peter K. <pe...@as...>]

On Mon, 10 Jan 2005, Thomas Zell wrote:
> However, I would suggest you try the Sipgate support, as it is quite likely a
> problem with _their_ configuration ...

I tried with free world dialup now, and got the same symptoms. I guess
I 'll make some more tests first. Doesn't sipgate use asterisk too?
Well, I guess there still has some work to be done before I have
my ubiquitous embedded-linux-powered ip mobile communicator...

-- 
peter koellner <pe...@as...>

Re: Antwort: [Siproxd-users] siproxd zyxel2000W problems

[Thomas Z. <t....@gm...>]

On Monday 10 January 2005 18:17, Mat...@mw... wrote:
> This does not seem to be a problem with your phone. I have exactly the
> same problem with the xten softphone and sipgate, but not with 1und1:
> I can call the 10000 number - ok
> I can be called with my sipgate number from outside
> But I cannot place a call. Get the error message from sipgate.
> No problems with 1und1 with the same configuration.
I also could not get Siproxd to work with Sipgate (with exactly the same=20
symptoms). I tried to figure out for a _very_ long time why that is the cas=
e=20
and I think I narrowed it down to the following: Siproxd adds a second VIA=
=20
header to every SIP Message it sends out (just as it is required by the=20
protocol) and Sipgate somehow has a problem with that. Sending out an INVIT=
E=20
with just one VIA header worked fine all the time, whereas exactly the same=
=20
message with 2 VIA headers always failed. Personally, I gave up at that poi=
nt=20
and installed Asterisk on my router.

However, I would suggest you try the Sipgate support, as it is quite likely=
 a=20
problem with _their_ configuration ...

=2D-=20
Thomas Zell

GPG Key: 2F528D7184975F94
=46ingerprint: 363A 7DA7 AAAD FD5D B1E2 4246 2F52 8D71 8497 5F94
sip: 53...@si... or +490221355337678

Antwort: [Siproxd-users] siproxd zyxel2000W problems

[<Mat...@mw...>]

This does not seem to be a problem with your phone. I have exactly the 
same problem with the xten softphone and sipgate, but not with 1und1:
I can call the 10000 number - ok
I can be called with my sipgate number from outside
But I cannot place a call. Get the error message from sipgate.
No problems with 1und1 with the same configuration.

Mathias Wohlfarth EDV-Beratung
Thomas-Mann-Str.1
53111 Bonn
Tel.    0172 / 53 45 591
        01801 / 777 555 33 01
Fax     0228 / 9469181
Email   mat...@mw...




Peter Koellner <pe...@as...>
Gesendet von: sip...@li...
10.01.2005 17:20
Bitte antworten an siproxd-users
 
        An:     Sip...@li...
        Kopie: 
        Thema:  [Siproxd-users] siproxd zyxel2000W problems


hi!

I got a Zyxel Prestige 2000W WLAN VOIP phone here that I am trying to get
to work with a sipgate account. Now the phones firmware is pretty 
primitive
with not much configuration options, and it did not work with the local
network setup, so I installed siproxd 0.5.9 (now the 0.5.10 snapshot as of
January 10th).
I am able to make calls to other softphones in the local net, but 
connections
to or from the outside fail with the following symptoms:

1. STUN ENABLED

-ringing the external phone works, ringing in from the external phone 
works,
  but audio does not work in either direction

2. STUN DISABLED

-ringing in works, with external audio source arriving, but the VOIP 
phones
audio source remains mute.
-ringing out to the landline phone gives an error message playback
  from sipgate
-calling sipgate's 10000 test number connects to the welcome message.

The network setup looks like this:

gateway:

Fedora Linux Box with kernel 2.6.7 and iptables NAT/firewall, 
ports 5060, 5061, 10000 and 7070:7079 open.
Q-DSL internet connection

local network: Netgear ME102 WLAN router

There also is an independent Asterisk installation on the gateway, so I 
use 
port 5061 for the outgoing proxy, but changing that to 5060 does not make 
any difference.

I am not too sure where to look for problems there. There are absolutely 
no
log messages from the phone itself, and I bet there are lots of bugs in
the firmware, but since local sip calls work I guess there must
be something going wrong with the RTP relaying. There is no restriction
on outgoing packets, so I do not think it is a firewall problem.

... and now I am totally confused because calls in the local network
do not work any more with the same configuration I tried last time...

Does anybody have any experience with the Zyxel phone? Or any suggestions
as how to pinpoint the problem?

-- 
peter koellner <pe...@as...>


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[Siproxd-users] siproxd zyxel2000W problems

[Peter K. <pe...@as...>]

hi!

I got a Zyxel Prestige 2000W WLAN VOIP phone here that I am trying to get
to work with a sipgate account. Now the phones firmware is pretty primitive
with not much configuration options, and it did not work with the local
network setup, so I installed siproxd 0.5.9 (now the 0.5.10 snapshot as of
January 10th).
I am able to make calls to other softphones in the local net, but connections
to or from the outside fail with the following symptoms:

1. STUN ENABLED

-ringing the external phone works, ringing in from the external phone works,
  but audio does not work in either direction

2. STUN DISABLED

-ringing in works, with external audio source arriving, but the VOIP phones
audio source remains mute.
-ringing out to the landline phone gives an error message playback
  from sipgate
-calling sipgate's 10000 test number connects to the welcome message.

The network setup looks like this:

gateway:

Fedora Linux Box with kernel 2.6.7 and iptables NAT/firewall, 
ports 5060, 5061, 10000 and 7070:7079 open.
Q-DSL internet connection

local network: Netgear ME102 WLAN router

There also is an independent Asterisk installation on the gateway, so I use 
port 5061 for the outgoing proxy, but changing that to 5060 does not make 
any difference.

I am not too sure where to look for problems there. There are absolutely no
log messages from the phone itself, and I bet there are lots of bugs in
the firmware, but since local sip calls work I guess there must
be something going wrong with the RTP relaying. There is no restriction
on outgoing packets, so I do not think it is a firewall problem.

... and now I am totally confused because calls in the local network
do not work any more with the same configuration I tried last time...

Does anybody have any experience with the Zyxel phone? Or any suggestions
as how to pinpoint the problem?

-- 
peter koellner <pe...@as...>

Re: [Siproxd-users] siproxd and cygwin

[Thomas R. <tr...@gm...>]

Hi Mathias,

Nice to hear that.

I Just put in tha Cygwin patch you sent. By the way: the ./configure
script is generated out of configure.in by autoconf.
Can you check that the current siproxd snapshot does build properly
with Cygwin?

Thanks,
/Thomas

On  7 Jan, Mat...@mw... wrote:
> We have installed and tested siproxd on Win2000 and WinXP with cygwin. 
> Runs stable.
> We have made the following changes, shown at the end of the mail:
> In configure we have introduced cygwin.
> In rtpproxy_relay.c we have exclude checking for uid 0, because with 
> cygwin there is no root user with uid 0. You can change in /etc/passwd, 
> but this is unsecure, because it is generated from the Windows users and 
> after generation random uids are found for the users.
> To install siproxd as a Windows service you can use
> cygrunsrv -I siproxd -p /usr/local/sbin -c /tmp -t auto -o -a "<parameters 
> for siproxd>"
> If you have cygwin init installed as service you can start siproxd from 
> inittab like in other Unix
> compile of libosip2-2.0.6 also worked fine: configure - make - make 
> install - ok.
> all good work
> regards MW
> here are the changes.
> configure:
> $ diff -c3 configure.orig configure
[snipp]
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VoIP:  sip:174...@pr... | sip:43...@fw...

[Siproxd-users] siproxd and cygwin

[<Mat...@mw...>]

We have installed and tested siproxd on Win2000 and WinXP with cygwin. 
Runs stable.
We have made the following changes, shown at the end of the mail:
In configure we have introduced cygwin.
In rtpproxy_relay.c we have exclude checking for uid 0, because with 
cygwin there is no root user with uid 0. You can change in /etc/passwd, 
but this is unsecure, because it is generated from the Windows users and 
after generation random uids are found for the users.
To install siproxd as a Windows service you can use
cygrunsrv -I siproxd -p /usr/local/sbin -c /tmp -t auto -o -a "<parameters 
for siproxd>"
If you have cygwin init installed as service you can start siproxd from 
inittab like in other Unix
compile of libosip2-2.0.6 also worked fine: configure - make - make 
install - ok.
all good work
regards MW
here are the changes.
configure:
$ diff -c3 configure.orig configure
*** configure.orig      Mon Jan  3 16:44:49 2005
--- configure   Mon Jan  3 16:47:11 2005
***************
*** 3313,3318 ****
--- 3313,3328 ----
  echo "$as_me:$LINENO: checking target platform" >&5
  echo $ECHO_N "checking target platform... $ECHO_C" >&6
  case "$target" in
+   *cygwin*)
+     echo "$as_me:$LINENO: result: Cygwin" >&5
+ echo "${ECHO_T}Cygwin" >&6
+
+ cat >>confdefs.h <<\_ACEOF
+ #define _CYGWIN
+ _ACEOF
+
+     ;;
+ case "$target" in
    *-*-linux*)
      echo "$as_me:$LINENO: result: Linux" >&5
  echo "${ECHO_T}Linux" >&6

rtpproxy_relay.c:
$ diff -c3 rtpproxy_relay.c.orig rtpproxy_relay.c
*** rtpproxy_relay.c.orig       Thu Dec 30 12:26:05 2004
--- rtpproxy_relay.c    Thu Dec 30 13:53:05 2004
***************
*** 111,122 ****
--- 111,124 ----
        int uid,euid;
        struct sched_param schedparam;

+ #ifndef _CYGWIN
        uid=getuid();
        euid=geteuid();
        DEBUGC(DBCLASS_RTP,"uid=%i, euid=%i", uid, euid);
        if (uid != euid) seteuid(0);

        if (geteuid()==0) {
+ #endif
  #if defined(HAVE_SCHED_GET_PRIORITY_MAX) && 
defined(HAVE_SCHED_GET_PRIORITY_MI
N)
           int pmin, pmax;
           /* place ourself at 1/3 of the available priority space */
***************
*** 134,144 ****
--- 136,148 ----
           if (sts != 0) {
              ERROR("pthread_setschedparam failed: %s", strerror(errno));
           }
+ #ifndef _CYGWIN
        } else {
           INFO("Unable to use realtime scheduling for RTP proxy");
           INFO("You may want to start siproxd as root and switch UID 
afterwards
");
        }
        if (uid != euid)  seteuid(euid);
+ #endif
     }
  #endif

Mathias Wohlfarth EDV-Beratung
Thomas-Mann-Str.1
53111 Bonn
Tel.    0172 / 53 45 591
        01801 / 777 555 33 01
Fax     0228 / 9469181
Email   mat...@mw...

[Siproxd-users] siproxd Authentication!

[<os...@gi...>]

Hello!

I'm having trouble authenticating users in siproxd! Siproxd always return=
s
"Proxy Authentication Required"!

I think I'm not configuring the authentication realm correctly!

Can anyone give me a hint in how to make this work?

Thanx!

=D3scar


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[Siproxd-users] siproxd-0.5.9 released

[Thomas R. <tr...@gm...>]

Release Notes for siproxd-0.5.9
===============================

Major changes since 0.5.8:
   Several bugfixes and better support for Record-Route headers.

General Overview:
 - SIP (RFC3261) Proxy for SIP based softphones hidden behind a
   masquerading firewall
 - works with "dial-up" conenctions (dynamic IP addresses)
 - Multiple local users/hosts can be masqueraded simultaneously
 - Access control (IP based) for incoming traffic
 - Proxy Authentication for registration of local clients (User Agents)
   with individual passwords for each user
 - May be used as pure Outbound proxy (registration of local UAs
   to a 3rd party registrar)
 - Fli4l OPT_SIP (still experimental) available, check
   http://home.arcor.de/jsffm/fli4l/
 - supports Linux, FreeBSD and Solaris
 - Full duplex RTP data stream proxy for *incoming* and *outgoing*
   audio data - no firewall masquerading entries needed
 - Port range to be used for RTP traffic is configurable
   (-> easy to set up apropriate firewall rules for RTP traffic)
 - RTP proxy can handle multiple RTP streams (eg. audio + video)
   within a single SIP session.
 - Symmetric RTP support
 - Symmetric SIP signalling support
 - Supports running in a chroot jail and changing user-ID after startup
 - All configuration done via one simple ascii configuration file
 - Logging to syslog in daemon mode
 - RPM package support
 - The host part of UA registration entries can be masqueraded
   (mask_host, masked_host config items). Some Siemens SIP phones seem to
   need this 'feature'.

Requirements:
 - pthreads (Linux)
 - glibc2 / libc5 / uClibc
 - libosip2

Currently tested on:
- Fedora Core1 (Kernel 2.4.x, Glibc)
  This is my main development and testing environment. Other platforms
  are not extensively tested by myself.

Builds on:
- Linux:	Fedora Core1
		WRT54g (133mhz mipsel router)
- FreeBSD:	FreeBSD 4.10-BETA
- OpenBSD:	OpenBSD 3.4 GENERIC#18
- SunOS:	SunOS 5.9
- Mac OS X:	Darwin 6.8

Reported interoperability with softphones:
 - Grandstream BudgeTone-100 series
 - Linphone (local and remote UA) (http://www.linphone.org)
 - Kphone (local and remote UA) (http://www.wirlab.net/kphone/)
 - MSN messenger 4.6 (remote and local UA)
 - X-Lite (Win XP Professional)
 - SJPhone softphone

Reported interoperability with SIP service providers:
 - Sipphone	(http://www.sipphone.com)
 - FWD		(http://www.fwd.pulver.com)
 - Sipgate	(http://www.sipgate.de)
 - Stanaphone	(SIP Gateway to PSTN)

 If you have siproxd successfully running with another SIP phone
 and/or service provider, please drop me a short note so I can update
 the list.

Known bugs:
   - SRV DNS records are not yet looked up, only A records
   There will be more...

If you port siproxd to a new platform or do other kinds of changes
or bugfixes that might be of general interest, please drop me a
line. Also if you intend to include siproxd into a software
distribution I'd be happy to get a short notice.


-----
md5sum for siproxd-0.5.9.tar.gz: 	7428bc04eb8d60a5741d68190b06f10b

GnuPG signature for siproxd-0.5.9.tar.gz archive:
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ChangeLog:
0.5.9
=====
  03-Nov-2004:  - Released 0.5.9
  31-Oct-2004:  - fix: A negative response to an INVITE shall stop
                  any associated initiated RTP streams
  24-Oct-2004:  - Default Expires timeout is now configurable.
  23-Oct-2004:  - fix: route_processing.c - when adding my record-route header,
                  the 'lr' parameter was not dynamically allocated memory
                  (possible crash when trying to free is)
                - up to 1000 characters per line in config file (was 120),
                  some typos corrected (by Tero Pelander)
                - fix: various correction in Record-Route processing
  13-Oct-2004:  - utils.c: preparation for chroot() (consider syslog)
  09-Oct-2004:  - included startup script (by Guido Trentalancia)
                - siproxd.spec: create PID and registrations directories
                  and install startup script


-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:174...@pr... | sip:43...@fw...