Hi All,
We are working to get our script working on Version 1.4 and are having
problems debugging the script. Previously if I saw the following error in
the logs it would highlight the line and colomb numbers were the error
existed. Can someone help out?
Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file,
line 9459656, column 1-1: syntax error
Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file,
line 9459656, column 1-1: bad command!)
Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors)
Thanks,
Jim
Central Time is not s valid timezone
Further than this your rates look ok at first glance, if it does not
work maybe other can help with their experiences.
Adrian
Hello everybody.
A patch has been published on tracker, which fixes a problem related to not
define a macro which makes fail the module compilation in Solaris.
It has been registered with Request ID: 2034302*
*
Hope it be useful.
Best regards.
Sergio Gutiérrez.*
*
I am afraid that rating all calls the same regardless of destination
will not work unless you define some destination to match all your
traffic.
The rating engine must find a destination prefix from which it derives
the price. Read the RATING.txt to understand the logic behind it.
Adrian
>>
I'm attempting to understand how call rating works within CDRTool. I am
able to link destinations to the calls, but unable to generate
prices. I'm
using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm
trying to setup seems simple: I have one set rate of $.08/min for any
call
beginning with a 1 (I'll exclude 800 numbers later). This rate should
be
applied globally with no differentiation between caller party or
destination...
Yes
You need to add at list one customer entry, associated profiles and
finally the rates linked with the profile which contain the actual
prices for each destination.
You have example rating files that you can modify.
Adrian
On Jul 31, 2008, at 3:28 PM, Brian Del Shasta wrote:
> That's not a problem. I am currently able to successfully match
> destinations by looking for anything that starts with a '1'. I have
> a destination setup called "Billable" which applies to any
> destination beginning with 1. I also have destinations setup for
> 1800, 1877, etc to filter out the 800 numbers.
>
> What I cannot figure out is how to link a rate to a destination. Do
> I have to setup a profile and customer as well? Or should I be able
> to price calls using only destination and rates?
>
> On Thu, Jul 31, 2008 at 8:42 AM, Adrian Georgescu <ag@ag-
> projects.com> wrote:
> I am afraid that rating all calls the same regardless of destination
> will not work unless you define some destination to match all your
> traffic.
> The rating engine must find a destination prefix from which it derives
> the price. Read the RATING.txt to understand the logic behind it.
> Adrian
> >>
> I'm attempting to understand how call rating works within CDRTool.
> I am
> able to link destinations to the calls, but unable to generate
> prices. I'm
> using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What
> I'm
> trying to setup seems simple: I have one set rate of $.08/min for any
> call
> beginning with a 1 (I'll exclude 800 numbers later). This rate should
> be
> applied globally with no differentiation between caller party or
> destination...
>
>
>
> _______________________________________________
> Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>
Hi Neill,
The outgoing audio works fine. The audio that is not passed is from PSTN -> Phone.
The Snom does have symmetrical RTP set. I did have this working about a week ago then it stopped working!
How does asterisk deal with NAT, as if I connect the phone directly to Asterisk everything works perfectly!
Both Asterisk & Kamailio are on publically accessible IP addresses.
Thank you for your help so far.
Ross
From: neill.wilkinson(a)btinternet.comTo: users(a)lists.kamailio.orgDate: Thu, 31 Jul 2008 13:32:05 +0100Subject: Re: [Kamailio-Users] NAT Problems
Which direction is the one way audio in – from the Phone to the PSTN or Vice-Versa – I assume audio outbound is OK and it’s the audio returning to the phone from the PSTN you’re missing?
Do you have symmetrical RTP set in the Line settings (under NAT) on the SNOM?
Is Kamailio inside or outside the NAT?
Neill...;o)
Neill WilkinsonPrincipal Consultant
Aeonvista Ltd - opening up new ideas
From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ross BeerSent: 31 July 2008 12:50To: users(a)lists.kamailio.orgSubject: [Kamailio-Users] NAT Problems
I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= if (!mf_process_maxfwd_header("10")) {sl_send_reply("483","Too Many Hops");exit;};if (msg:len >= 2048 ) {sl_send_reply("513", "Message too big");exit;};# NAT detectionforce_rport();if(nat_uac_test("3")){fix_nated_contact();}if (!method=="REGISTER"){record_route();}if (loose_route()) {append_hf("P-hint: rr-enforced\r\n"); route(1);};if (!uri==myself) {append_hf("P-hint: outbound\r\n"); route(1);};if (uri==myself) {if (method=="REGISTER") {if (!www_authorize("", "subscriber")) {www_challenge("", "0");exit;};if (isflagset(5)) {# set branch flag -- when someone will call this user# the INVITE will have branch flag 6 set after lookup("location")setbflag(6); };fix_nated_contact();fix_nated_register();consume_credentials();save("location");exit;};if (!lookup("location")) {sl_send_reply("404", "Not Found");exit;};append_hf("P-hint: usrloc applied\r\n"); };if(method=="MESSAGE"){if (!lookup("location")){sl_send_reply("404", "User Offline");exit;}route(4);}if (method=="REGISTER"){save("location");} route(1);} route[1] {if (subst_uri('/(sip:.*);nat=yes/\1/')){setbflag(6);};if (isflagset(5)||isbflagset(6)) {route(3);}t_on_reply("1");if (!t_relay()) {sl_reply_error();};exit;} route[3] {if (is_method("BYE")) {unforce_rtp_proxy();} else if (is_method("INVITE")){fix_nated_sdp("3"); };if (isflagset(5)){search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');}t_on_reply("1");}route[4]{if (!t_relay()) {sl_reply_error();};exit;} failure_route[2] {if (isbflagset(6) || isflagset(5)) {unforce_rtp_proxy();}}onreply_route[1] {fix_nated_contact();if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) {fix_nated_sdp("2");force_rport(); }search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');} ==================== SIP PACKETS ===========================INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asteriskMax-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxxs=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY ////////////////////SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID ***CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20
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I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio.
I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be.
Outgoing calls from Kamailio -> asterisk -> PSTN work fine.
Thank you for your help in advance, it is much appreciated.
=================== config =======================
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# NAT detection
force_rport();
if(nat_uac_test("3"))
{
fix_nated_contact();
}
if (!method=="REGISTER")
{
record_route();
}
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
};
if (uri==myself) {
if (method=="REGISTER")
{
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
};
if (isflagset(5))
{
# set branch flag -- when someone will call this user
# the INVITE will have branch flag 6 set after lookup("location")
setbflag(6);
};
fix_nated_contact();
fix_nated_register();
consume_credentials();
save("location");
exit;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
if(method=="MESSAGE")
{
if (!lookup("location"))
{
sl_send_reply("404", "User Offline");
exit;
}
route(4);
}
if (method=="REGISTER")
{
save("location");
}
route(1);
}
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')){
setbflag(6);
};
if (isflagset(5)||isbflagset(6)) {
route(3);
}
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[3] {
if (is_method("BYE"))
{
unforce_rtp_proxy();
}
else if (is_method("INVITE"))
{
fix_nated_sdp("3");
};
if (isflagset(5))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
}
t_on_reply("1");
}
route[4]
{
if (!t_relay())
{
sl_reply_error();
};
exit;
}
failure_route[2] {
if (isbflagset(6) || isflagset(5))
{
unforce_rtp_proxy();
}
}
onreply_route[1]
{
fix_nated_contact();
if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE"))
{
fix_nated_sdp("2");
force_rport();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
}
==================== SIP PACKETS ===========================
INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asterisk
Max-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxx
s=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv
////////////// PHONE REPLY ////////////////////
SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID ***
CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508
v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***
t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20
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