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Hi guys,
I've been experiencing some troubles with carrier's that are using
Nextone (www.nextone.com).
I could place calls through Nextone but no longer then 1 minute, between 48 and
53 seconds the call is dropped with a BYE sent by Nextone.
I've been using another carriers using Cisco and Quintum without problems.
Does some one experimenting this problem?
I'm using SER 0.8.14.
Thanks in advance.
- --
============================================
Rodrigo P. Telles <telles(a)devel.it>
IVOZ # 1009
Project Manager
Devel-IT - http://www.devel.it
Bestcom Group
============================================
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Hi,
With the advent of TLS support for SER, i was testing it with KPhone
(TLS being used between SER proxies only, Kphone-to-SER via UDP).
Turned out that kphone would not respond to any message (it would
receive them, but plain ignore them).
The problem: TLS as transport mode in the VIA headers was not
supported,
thus, the SIP messages were considered not valid.
I am attaching a patch to correct this for kphone 4.1.0, but i guess it
can be applied
with little modifications to earlier versions (4.0.3 to 4.0.5) but i am
not sure.
Regards,
Cesc
PATCH STARTS HERE:
Index:
dissipate2/sipclient.cpp===================================================================---
dissipate2/sipclient.cpp (revision 1)+++
dissipate2/sipclient.cpp (working copy)@@ -1034,6 +1034,9 @@
printf( "SipClient: Sending TCP
Response\n" ); outsocket = new TCPMessageSocket;
break;+ case
SipVia::TLS:+ printf( "SipClient: TLS in
top via, not supported (full TLS support not implemented)\n"
);+ break; case
SipVia::BadTransport: printf( "SipClient: Bad
transport on incoming Via\n" ); break;@@ -1232,6
+1235,9 @@ printf( "SipClient: Sending TCP
Response\n" ); outsocket = new TCPMessageSocket;
break;+ case
SipVia::TLS:+ printf( "SipClient: TLS in
top via, not supported (full TLS support not implemented)\n"
);+ break; case
SipVia::BadTransport: printf( "SipClient: Bad
transport on incoming Via\n" ); break;Index:
dissipate2/sipvia.h===================================================================---
dissipate2/sipvia.h (revision 1)+++ dissipate2/sipvia.h (working
copy)@@ -49,6 +49,7 @@ enum Transport { UDP,
TCP,+ TLS, BadTransport };
/**Index:
dissipate2/sipvia.cpp===================================================================---
dissipate2/sipvia.cpp (revision 1)+++ dissipate2/sipvia.cpp (working
copy)@@ -37,6 +37,7 @@ switch ( t ) { case UDP: return "UDP";
case TCP: return "TCP";+ case TLS: return
"TLS"; case BadTransport: return "BAD"; } return
QString::null;@@ -46,6 +47,7 @@ { if ( t.compare(
getTransportString( UDP ) ) == 0 ) { return UDP; } if ( t.compare(
getTransportString( TCP ) ) == 0 ) { return TCP; };+ if ( t.compare(
getTransportString( TLS ) ) == 0 ) { return TLS; }; return
BadTransport; }
PATCH ENDS HERE
Unclassified
Hi serusers,
Anybody has a sample ser.cfg that forward calls to Cisco AS5300 GW?
I am using the Nathelper templet config file and got NAT +RTP relay works on IP2IP, but dont know how to work with PSTN through Cisco AS5300 GW.
where to put the forwording code.
I got some sample code here:
# is it a PSTN destination? (is username nummerical and does not begin with 8?)
if (uri=~"^sip:[0-79][0-9]*@") { # ... forward to gateways then;
# check first to which PSTN destination the requests goes;
# if it is US (prefix "1"), use the gateway XXX.XXX.XXX.XXX...
if (uri=~"^sip:1") {
# strip the leading "1"
strip(1);
forward(XXX.XXX.XXX.XXX, 5060);
} else {
# ... use the gateway YYY.YYY.YYY.YYY for all other destinations
forward(YYY.YYY.YYY.YYY , 5060);
}
break;
}
I dont know if it can talk with Cisco 5300 GW though.
Where should I put it in ser.cfg with NAThelper + RTPreply support?
How to debug the ser.cfg, if i hear busy tone when i dial the PSTN number.
Thanks for your time,
Steve
---------------------------------
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Yahoo! Small Business - Try our new resources site!
Hi Greger,
The system is currently being tested by someone else
but I believe they are behind a Linksys VPN router.
Are you suggesting it could simply be the settings in
this?
I "think" I understand the nat issues associated with
sip and sdp fairly ok so would I be correct in saying
that if my two clients are behind nat(the same nat)on
the same subnet the rtpproxy should be invoked? This
would be my understanding of the situation but then I
saw a recent email (see message header below)which
suggests an external script should be used.
Re: FW: [Serusers] calls between UA´s b ehind same NAT
us ing nathelper/rtpproxy
Also what confuses me is that the scenario works
sometimes and yet other times it doesnt. I will
attempt to get a full message dump (of both the
working and non working scenario)from the tester if
that will help.
Kindest Regards,
Pat.
--- "Greger V. Teigre" <greger(a)teigre.com> wrote:
> Pat,
> You haven't said anything of the type of NAT you are
> behind. To me it sounds
> like an ALG (Application layer gateway) problem. Try
> to turn of the SIP ALG
> in your router. If not, please post a full SIP
> message exchange. You need
> to find out if they communicate through the NAT
> (hairpin media) or directly.
> That depends on the SDP payload in the INVITE and OK
> messages.
> The new Getting Started document on
> http://onsip.org/ (you need to
> register) has a thorough review of NAT issues and
> rewriting. Recommend! (I
> wrote it ;-) )
> g-)
>
> pat newham wrote:
> > Following on from my below email, I can now
> definately
> > say the problem is not nat pings. Just to recap I
> am
> > experiencing intermittent audio. It works when the
> > phones have very recently registered, then
> sometimes
> > theres one way audio and then sometimes no audio.
> Does
> > anyone have any ideas what the problem could be or
> > where I could begin to troubleshoot this?
> >
> > Hi,
> >
> > I have a strange problem. I have two grandstream
> > budgetone clients on the same subnet behind nat
> > registering with ser on a public address.
> Obviously
> > their public addresses would be the same but they
> > listen on different ports. When they initially
> > register, I can the call,audio is transmitted and
> > everything is successful.
> >
> > However sometimes theres only one way audio, other
> > times theres no audio and then other times it
> > works....I am guessing that this is because the
> nat
> > router is forgetting the nat mapping so after a
> while
> > when the nat mapping is "forgotten" and a packet
> > arrives destined for a client, the router drops
> it....
> >
> > Could someone verify this for me??...Am I on the
> right
> > track?? I have the following settings in ser.cfg
> which
> > I thought would keep the nat settings alive.
> >
> > modparam("registrar", "nat_flag", 6)
> > modparam("nathelper", "natping_interval", 30) #
> Ping
> > interval 30 s
> > modparam("nathelper", "ping_nated_only", 1) #
> Ping
> > only clients behind NAT
> >
> > I also increased the nat keep alives "pings" sent
> in
> > the configuration settings of the grandstream
> > phone....Any further ideas??
> >
> > Regards,
> > Pat.
> >
> > Send instant messages to your online friends
> > http://uk.messenger.yahoo.com
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
Send instant messages to your online friends http://uk.messenger.yahoo.com
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Hi folks,
I'm really newbie on SER, but I've a project in my mind, and I would
your advices and suggestions, if you please.
Situation: I've multiple SIP accounts, and my Cisco CME could register
only one with sip-ua and a dial-peer voip. And the Cisco CME hasn't
itself a voicemail solution.
So, my idea is: register the SIP accounts on the SER, then forward all
calls to an internal SIP number that I'll register to CME. Then, if I
need a voicemail, I could forward calls from CME to SER (there's a
voicemail feature? could I try to record the call and play back with my
ip phone and/or send the wav file as attach via email?).
I hope you could put me in the right direction.
Thanks for any support, question or advice.
Regards
Andrea
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What's the meaning of the state field in the location table?. Usually
the value is 0. But sometimes, its value is 1 and then we have some
problems in our applications. What's the difference?.
Thanks
hello group,
has anyone experience with ser's enum module and tried it with e164.org?
i've something like in my ser.cfg:
if(enum_query("e164.org")){
....
};
i tried to dial a number which is known there but I won't get into this
if clause.
but when i try
if(enum_query("e164.arpa")){
....
};
and call a number known there the enum_query works?
any help with enum would be greatly appreciated.
thanks in advance.
pamela
Hi:
I had a very basic doubt with respect to the function "hold/unhold" on ser
with rtpptoxy and kphone.
Do nathelper and rtpproxy currently support hold/unhold function (with
kphone 4.1.0) with nated clients ?
Or do they have other methods to acheive that function?
In my test, I put rtpproxy and ser in same pc, and two UACs are behind
that pc,
When I use hold/unhold, most time it works but some times it doesn't work
I checked the ethereal log, it seems that the fail happens only when
1. hold function last less than 60 seconds.
2. once the re-invite's rtp port in SDP announced by client has been changed
to different rtp port (different from original rtp port).
So it seems rtpproxy doens't knows the ports come from both UACs have
changed...
and didn't forward UACs' rtp packets.
Here is the question I have conclude from above description
1. While receive the re-invite message, will ser recheck the port in SDP
and announce rtpproxy again?
2. Is this the limitation of Ser+rtpproxy with NATed UACs?
3. Does there has any suggestion that can make this function works?
(ser+nathelper+rtpproxy with function hold/uhhold)
Thanks and best regards
Jimmy
Many people have reported having problems logging in to serweb after
installation.
I had the problem myself, and after some inspection of the
html/admin/index.php file, the following lines game me a hint:
if ($config->clear_text_pw) {
$q="select phplib_id from ".
$config->table_subscriber.
" where
username='".addslashes($uname)."' and password='".addslashes($passw)."'
and perms='admin' and domain='".addslashes($config->realm)."'";
So, the SELECT statament wants to match the 'username', 'password',
'perms' and 'domain' fields from the 'subscriber' table.
In your html/config.php file, comment-out the regex replace and type
your domain exactly as it is in your db.
e.g:
/* your domain name */
# $this->realm=$this->domainname=$this->default_domain=
ereg_replace( "(www\.|sip\.)?(.*)", "\\2", $_SERVER['SERVER_NAME']);
$this->realm="mydomain.com";
Hope this helps.
cv
