- sr-users - kamailio.org

[Serusers] How can I keep SER from changing reply Response Code from 503 to 500?
by Frank Durda IV 09 Oct '08

09 Oct '08

I have a situation where SER is gumming up the works. The problem is that I have a call where the INVITE fails because a PSTN gateway hasn't authorized the SIP invitee to call a given number, as intended. The PSTN switch is returning a "503 Service Unavailable" as is expected, but SER is then sending the SIP invitee a "500 Service Unavailable" instead of the original message, which is wrong in several ways. Here is the ngrep of what SER received from the PSTN switch (which is the desired answer in this situation): # U 2008/10/09 18:08:45.045015 10.133.0.2:5060 -> 10.133.90.6:5060 SIP/2.0 503 Service Unavailable. Call-ID: 14407805-3432564525-49719(a)sbc1.sometelco.com. CSeq: 1 INVITE. From: <sip:8019995678@209.x.x.x>;tag=3432564525-49731. To: <sip:9179891234@208.x.x.x:5060;user=phone>;tag=000a0285+1+4bae000a+de6c6831. Via: SIP/2.0/UDP 10.133.90.6;branch=z9hG4bKaa0f.96addb37.0. Via: SIP/2.0/UDP 209.x.x.x:5060;branch=z9hG4bKafec84c056d5d9ac97d27cc12d3dbb92. Server: DC-SIP/2.0. Content-Length: 0. Retry-After: 3. . That response code (503 Service Unavailable) is what should be sent back to the call originator, who can cope with it and redirect the call. However, here is the ngrep of what SER transmitted on to the invitee instead: # U 2008/10/09 18:08:45.050440 10.133.90.6:5060 -> 209.x.x.x:5060 SIP/2.0 500 Service Unavailable. To: <sip:9179891234@208.x.x.x:5060;user=phone>;tag=b11e68d9aa35b34c54f995457fe194f9-0069. From: <sip:8019995678@209.x.x.x>;tag=3432564525-49731. Call-ID: 14407805-3432564525-49719(a)sbc1.sometelco.com. CSeq: 1 INVITE. Via: SIP/2.0/UDP 209.x.x.x:5060;branch=z9hG4bKafec84c056d5d9ac97d27cc12d3dbb92. Server: Sip EXpress router (2.0.0-rc1 (x86_64/freebsd)). Content-Length: 0. . And the calling party receives the confusing "503 Service Unavailable". The onreply_route routine in use is a strip-down of what is found in the PSTN sample (page 54 of the SER Getting Started manual), and it makes no attempt to alter response codes: onreply_route["PSTN_REPLY"] { xlog("L_ERR","In onreply_route %rs %rr"); force_send_socket("10.133.90.6"); #emit on em1 fix_nated_contact("208.x.x.x"); /*Fixes Contact:*/ # Apply RTP proxy if necessary, but only for INVITE transactions # and 183 or 2xx replies if (@cseq.method != "INVITE") return; xlog("L_ERR","onreply_route[INVITE_REPLY] R1 %rs %rr"); if ((status =~ "(183)|2[0-9][0-9]") && search("^(Content-Type|c):.*application/sdp")) { xlog("L_ERR","In onreply_route FIXING PROXY"); force_rtp_proxy("e","208.x.x.x"); } return; } From debug messages (show below), the call to force_rtp_proxy() is not being performed, so in theory this routine is left with the 503 still intact. The xlog message shows the 503 value was still there on entry to onreply_route as well as near the end of the routine: Oct 9 18:08:45 ser1 /usr/local/sbin/ser[1820]: In onreply_route 503 Service Unavailable Oct 9 18:08:45 ser1 /usr/local/sbin/ser[1820]: onreply_route[INVITE_REPLY] R1 503 Service Unavailable The "FIXING PROXY' debug message was not reached, as expected. So what is causing: 503 Service Unavailable. to mysteriously turn into 500 Service Unavailable. which isn't even the correct RFC textual response for a 500 code. This is being seen in ser-2.0.0-rc1 and is completely repeatable. Ideas? Suggestions? Thanks in advance!

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[Serusers] serctl moni problems
by psycho psycho 09 Oct '08

09 Oct '08

Hi, I have a problem with the serctl utility. Even though the ser process starts without any problems, when i am trying to use the monitor by executing serctl moni i get the following error [: 1074: 0: unexpected operator. All seems to work (making call etc) but monitor. If the number 1074 points to a line from the serctl then i attach is my serctl source (i don't know if that helps): 1073 cisco_restart) 1074 if [ "$#" -ne 2 ] ; then 1075 usage 1076 exit 1 1077 fi Any suggestions.? Thank you for your time. _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

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[Kamailio-Users] Problem with call on hold
by Diego Zuaneti Arruda 09 Oct '08

09 Oct '08

Hi all, I'm having a problem when the user places the call on hold. When the user removes the call on hold, only one side hears the call. Does anyone know what is wrong? The dialog of the call is attached. Thanks. Diego.

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Re: [Kamailio-Users] Users Digest, Vol 41, Issue 24
by luzango mfupe 09 Oct '08

09 Oct '08

Ok David, will give feedback soon.rgds, Lu. On Thu, Oct 9, 2008 at 11:53 AM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: No audio both ways. (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 9 Oct 2008 11:07:12 +0200 > From: "David Villasmil" <david.villasmil.work(a)gmail.com> > Subject: Re: [Kamailio-Users] No audio both ways. > Cc: users(a)lists.kamailio.org > Message-ID: > <9853f4ff0810090207p11d15bddt8dca8f57370c83c9(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Try adding a: > > nat=yes > > to the kamailio/openser peer definition and test > > dvg > > On Thu, Oct 9, 2008 at 10:58 AM, luzango mfupe <luzango.mfupe(a)gmail.com > >wrote: > > > > > Hi mates,I have this setup: > > Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider > > > > I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port 5065) > on > > the same Debian Box in Realtime with no NAT. Asterisk connects calls to > the > > VoIP to PSTN provider. I am able to establish calls towards the PSTN > > side(Landline & Mobiles) but with no audio. I can hear the ringing tone > but > > when the call connects and the conversation begin i hear nothing so as > the > > Callee side. > > > > Below are my configs,the ngrep captured packets and codecs. > > > > > #################################################################################### > > route[4] { > > # routing to the public network > > rewritehostport("xx.xxx.xxx.xx:5065"); > > t_on_failure("2"); > > if (!t_relay()) { > > sl_reply_error(); > > }; > > exit; > > } > > > > route[6] { > > # > > # -- NAT handling -- > > # > > if (isbflagset(6) || isbflagset(7)) { > > append_hf("P-hint: Route[6]: mediaproxy \r\n"); > > use_media_proxy(); > > }; > > } > > > > route[10] { > > #from an internal domain -> inbound > > #Native SIP destinations are handled using the location table > > #Gateway destinations are handled by regular expressions > > append_hf("P-hint: inbound->inbound \r\n"); > > > > if (uri=~"^sip:0[1-9][0-9]+@.*") { > > if (is_user_in("credentials","local")) { > > strip(1); > > prefix("27"); > > route(6); > > route(4); > > exit; > > } else { > > sl_send_reply("403", "No permissions for local calls"); > > exit; > > }; > > }; > > > > if (uri=~"^sip:00[1-9][0-9]+@.*") { > > if (is_user_in("credentials","int")) { > > strip(2); > > route(6); > > route(4); > > exit; > > } else { > > sl_send_reply("403", "No permissions for international > calls"); > > }; > > }; > > > > > > > ################################################################################### > > This call was from the xlite softphone 1974 towards the Landline > > 0123825710. > > > > > ################################################################################### > > U 2008/12/06 03:38:43.896057 196.212.209.18:46738 -> kamailio IP:5060 > > INVITE sip:0123825710@KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP > 192.168.0.55 > > : > > > > > 46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward > > s: 70..Contact: <sip:1974@196.212.209.18:46738>..To: > > "0123825710"<sip:01238 > > 25710@kamailio IP>..From: <sip:1974@kamailioIP>;tag=353dd217..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Allow: > > INVIT > > E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, > > INFO..Cont > > ent-Type: application/sdp..User-Agent: X-Lite release 1014k stamp > > 47051..Co > > ntent-Length: 237....v=0..o=- 1 2 IN IP4 192.168.0.55..s=CounterPath > > X-Lite > > 3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3 > > 101..a=fmtp > > :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 : ljiQYRpD > > NiMZsfdZ > > 192.168.0.55 60782..a=sendrecv.. > > > > U 2008/12/06 03:38:43.896350 Kamailio IP:5060 -> 196.212.209.18:46738 > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > 192.168.0.55:46 > > > > > 738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received > > =196.212.209.18..To: "0123825710"<sip:0123825710@kamailioIP > >;tag=329cfea > > a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@kamailioIP > >;tag=353dd217 > > ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 > > INVITE..Pr > > oxy-Authenticate: Digest realm="41.208.212.97", > > nonce="4939d8cfeb060ab14354 > > 85eee811cdf644f759a2"..Content-Length: 0.... > > > > U 2008/12/06 03:38:44.086313 196.212.209.18:46738 -> kamailio IP:5060 > > ACK sip:0123825710@kamailio IP SIP/2.0..Via: SIP/2.0/UDP > > 192.168.0.55:467 > > 38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To: > > "012382571 > > 0"<sip:0123825710@kamailioIP > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. > > .From: <sip:1974@41.208.212.97 <sip%3A1974(a)41.208.212.97> < > sip%3A1974(a)41.208.212.97 <sip%253A1974(a)41.208.212.97> > >>;tag=353dd217..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3 > > ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0.... > > > > > > > > U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio > IP;branch=z9hG4bKd78.7bd576 > > 24.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738 > > ;received=1 > > 96.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673 > > 8..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: > <si > > p:1974@kamailio IP>;tag=353dd217..To: > > "0123825710"<sip:0123825710@41.208. > > 212.97>..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2 > > INV > > ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, > BYE, > > RE > > FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: > > <sip:27123825710@41.2 > > 08.212.97:5065>..Content-Length: 0.... > > > > U 2008/12/06 03:38:44.711225 70.42.72.49:5060 -> Asterisk IP:5065 > > SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk > IP:5065;branch=z9hG4b > > K22abec12;rport=5065..From: "1974" <sip:1974@AsteriskIP > :5065>;tag=as6a7c > > b89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(a)70.42.72.49> > >>..Call-ID: > > 1934f5d443abffe07 > > c59d0a42215b49c@41.208.212.97..CSeq: 102 INVITE..Server: OpenSER > > (1.3.2-not > > ls (i386/solaris))..Content-Length: 0.... > > > > U 2008/12/06 03:38:47.206445 70.42.72.49:5060 -> Asterisk IP:5065 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk > IP:5065;branch=z9 > > hG4bK22abec12;rport=5065..From: "1974" <sip:1974@AsteriskIP > :5065>;tag=as > > 6a7cb89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(a)70.42.72.49> > > > > >;tag=cba-1a6e-48ecbab6.. > > Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 > > INVITE.. > > Contact: <sip:1214650027123825710@70.42.72.138<sip%3A1214650027123825710(a)70.42.72.138> > <sip%3A1214650027123825710(a)70.42.72.138<sip%253A1214650027123825710(a)70.42.72.138> > >>..Date: > > Wed, 08 Oct 2008 13: > > 50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route: <sip:70.42.72.49 > > ;lr=on;fta > > g=as6a7cb89f>..Content-Type: application/sdp..Content-Length: > > 212....v=0..o > > =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN IP4 > > 216.49 > > .201.22..t=0 0..m=audio 27852 RTP/AVP 0 101..a=ptime:20..a=rtpmap:0 > > PCMU/80 > > 00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. > > > > U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio > IP;branch=z9hG4bK > > d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP > 192.168.0.55:46738 > > ; > > received=196.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-; > > rport=46738..Record-Route: <sip:41.208.212.97 > > ;lr=on;ftag=353dd217;nat=yes>. > > .From: <sip:1974@kamilio IP>;tag=353dd217..To: > "0123825710"<sip:01238257 > > 10(a)41.208.212.97>;tag=as4377a96d..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl > > NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, > > ACK, > > CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: > > replaces..Conta > > ct: <sip:27123825710@Asterisk IP:5065>..Content-Type: > > application/sdp..Co > > ntent-Length: 287....v=0..o=root 2664 2664 IN IP4 > > 41.208.212.97..s=session. > > .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 > > 101..a=rtpmap:8 > > PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 > > telepho > > ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - > > -..a=ptime:20..a=se > > ndrecv.. > > > > U 2008/12/06 03:38:47.207106 Kamailio IP:5060 -> 196.212.209.18:46738 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.55:46738 > > ;received= > > 196.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467 > > 38..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: > <s > > ip:1974@Kamailio IP>;tag=353dd217..To: > > "0123825710"<sip:0123825710@kamilio IP>;tag=as4377a96d..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ > > jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, > > CANCEL, > > OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: > > <sip: > > 27123825710@41.208.212.97:5065>..Content-Type: > > application/sdp..Content-Len > > gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk IP..s=session..c=IN > IP4 > > 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 > > PCMA/800 > > 0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 > > telephone-event/ > > 8000..a=fmtp:101 0-16..a=silenceSupp:off - - - > > -..a=ptime:20..a=sendrecv.. > > > > > > > ######################################################################################## > > Sip.conf > > [general] > > context=from-trunk > > bindport=5065 > > autocreatepeer=yes > > bindaddr=xx.xxx.xxx.xx > > > > > > disallow=all > > ;allow=gsm > > ;allow=amr > > allow=alaw > > allow=ulaw > > allow=gsm > > ;allow=ilbc > > ;disallow=all > > ; > > ;useragent=Asterisk PBX > > dtmfmode = rfc2833 > > > > > > domain=xx.xxx.xxx.xx ; Add IP address as local domain > > > > [Provider] > > disallow=all > > canreinvite=no > > context=from-trunk > > allow=all > > ;allow=ulaw > > ;allow=gsm > > host=xx.xx.xx.xx > > insecure=port,invite > > type=peer ; we only want to call out, not > be > > call$ > > dtmfmode=rfc2833 > > > > > ######################################################################################### > > Here is my codecs > > > > 41*CLI> core show translation > > Translation times between formats (in milliseconds) for one > second > > of data > > Source Format (Rows) Destination Format (Columns) > > > > g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc > g726 > > g722 amr > > g723 - - - - - - - - - - - > - > > - - > > gsm - - 2 2 2 2 1 5 - 19 - > 2 > > - 14 > > ulaw - 5 - 1 2 2 1 5 - 19 - > 2 > > - 14 > > alaw - 5 1 - 2 2 1 5 - 19 - > 2 > > - 14 > > g726aal2 - 5 2 2 - 2 1 5 - 19 - > 1 > > - 14 > > adpcm - 5 2 2 2 - 1 5 - 19 - > 2 > > - 14 > > slin - 4 1 1 1 1 - 4 - 18 - > 1 > > - 13 > > lpc10 - 6 3 3 3 3 2 - - 20 - > 3 > > - 15 > > g729 - - - - - - - - - - - > - > > - - > > speex - 6 3 3 3 3 2 6 - - - > 3 > > - 15 > > ilbc - - - - - - - - - - - > - > > - - > > g726 - 5 2 2 1 2 1 5 - 19 - > - > > - 14 > > g722 - - - - - - - - - - - > - > > - - > > amr - 6 3 3 3 3 2 6 - 20 - > 3 > > - - > > > > > > > > With best regards, > > Lu. > > > > -- > > Luzango Mfupe > > TUUNE MOBILE > > Tel:0128440528/0123825710 > > Tshwane-RSA > > > > "...Ships are safe in harbor, but they were never meant to stay > > there......." > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > >

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[Serusers] SIP MESSAGE BODY ENCRYPTION
by Adeyeye Oluwasegun Michael 09 Oct '08

09 Oct '08

Hi, I was wondering if the content or body of a SIP MESSAGE from a SIP client is always encrypted when a TLS-supported SIP proxy for the client registration and the likes. I know that S-MIME supported SIP stack could do that for me, but I was wondering if a TLS-supported proxy could do as well. Thanks, Michael.

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[Kamailio-Users] SST module: Min-SE optional in request
by Schumann Sebastian 09 Oct '08

09 Oct '08

Dear all I might have noticed a bug in SST module: If I send an INVITE with Session-Expires: 300 Supported: timer I get rejected with 422 Session-Timer too small. If a Min-SE: x header with x > set min_se parameter included I do not get the message. This is IMHO wrong behavior, as Min-SE is optional in initial INVITE-requests sent by the phone acc. RFC 4028, table 1. Another thing that I noticed is the response to this message includes the Min-SE header (as it should), but has no linebreak after it (at least in kamailio branch 1.4). The message looks like MIN-SE:120Server: Kamailio (1.4.1-notls (i386/linux)) The phone call cannot be performed. At least one thing is IMHO wrong, as the phone cannot complete the call. It might be wrong the call gets rejected, or the answer might be possible like this, but then the phone cannot extract Min-SE due to wrong format. Any opinions on this? Sebastian

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[Kamailio-Users] !!Urgent Need to setup SIP SERVER like SIP Account Provider as my own Server.Thanks for your very great Help.
by sam 09 Oct '08

09 Oct '08

HI , I want to be a SIP providers like broadvoice gimzo ,where can i get DID numbers ,do we need to have contact with telecommunication Department. Please give me a full details.When we use *kamailio *,then there is no need to get sip account and virtual numbers. -- Cheers K.S.Sampath Kumar

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[Kamailio-Users] Kamailio and Asterisk trunking?
by John D 09 Oct '08

09 Oct '08

Not sure if I'm properly wording this, but I used to have two asterisk servers speaking to each other over SIP, configured via sip.conf. I now need a way to forward calls from my Asterisk instance to my Kamailio server so it can properly distribute the call. At this point however, I cannot find any documentation in order to properly have the two speak to each other. Would anyone be kind enough to point me in the right direction to establish that connection? Thanks.

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[Kamailio-Users] Attended Transfer with asterisk
by ALAEDDINE abbech 08 Oct '08

08 Oct '08

Hello, Does anybody got a working configuration to make an "attended call transfer" with a call through an Asterisk gateway? Example: PSTN --> Asterisk --> SER --+-- A |<---transfer call pstn to B +-- B The call will come from the PSTN Network and will go through "A". A sets the call on "Hold" and calls "B". After A is connected with B, A hangup and B got the call from PSTN. This is not working at the moment. Attended call transfer only with OpenSER and only with SIP-Phones is no Problem. But if the is an Asterisk as PSTN-GW in the game it will not work.

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[Kamailio-Users] ACK without matching transaction discarded
by Alex R.S.M 08 Oct '08

08 Oct '08

Hi, I am using openSER as a proxy for a simple SIP call (UAC <--> openSER <--> UAS). openSER receives ACK from callee but doesn't send it to caller. It seems openSER can not find a matching transaction, so it ignore and discard the message. if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } Here is the message header: Internet Protocol, Src: 10.0.0.174 (10.0.0.174), Dst: 10.0.0.160 (10.0.0.160) ... ACK sip:9134750000@10.0.0.183:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.174:5040;branch=z9hG4bK-12255-1-4 From: <sip:9133750000@10.0.0.174:5040>;tag=1 To: <sip:9134750000@10.0.0.183:5080>;tag=1 Call-ID: 1-12255(a)10.0.0.174 CSeq: 1 ACK Contact: sip:sipp@10.0.0.174:5040 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 Am I missing anything? Thanks, Alex _________________________________________________________________ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

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[Kamailio-Users] Ping both UACs when a call starts
by Cesar Troncoso 08 Oct '08

08 Oct '08

Hello! When a call starts I would like OpenSER to ping (by sending OPTIONS or INFO request) both UAC´s just to know if they are still there or for some reason they crashed or lost connectivity. In case they don't response, the call should be finished. What is the best way to do this? Thanks in advance! Cesar

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[Kamailio-Users] openser 1.3.3 domain module
by toly 07 Oct '08

07 Oct '08

Greetings, In domain module, command is_domain_local - using parameter_fixup where there is a call pkg_malloc, but nowhere function free_fixup is used. This will be memory leak, or I'm missing something. Thanks for the prompt reply, Toly -- View this message in context: http://www.nabble.com/openser-1.3.3-domain-module-tp19863638p19863638.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] LCR to Carrierroute with DNS/SRV Support
by Krom 07 Oct '08

07 Oct '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for URI manipulation prior to relaying to a cluster of session boarder controllers. As you know the DB table for LCR only supports integers in the gw table for ip_addr. I have been reading up on 1.2.x and 1.3.x to upgrade to a solution that would allow us to stay within the same call processing (script) design, but still support the new upstream requirement. I noticed the carrierroute table rewrite_host column is string. Could this be used to satisfy my needs of manipulating the URI for an IP where the carrier still uses a list of gateway IP's to try and use the DNS/SRV domain name where the carrier requires the use for SRV look-up? The session boarder controllers will conduct the SRV look-up after receiving the relayed message from Kamailio, it is just getting the domain name on the URI instead of IP address that I am looking to accomplish. Will the carrierroute module assist in this effort or is there another way that I am totally missing? Figured I would ask the gurus before following Alice down the rabbit hole. -- Thanks, -- Kyle

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[Kamailio-Users] Can openser.cfg lead to pkg memory problem?
by mayamatakeshi 07 Oct '08

07 Oct '08

Hello, we have openser 1.3.3 running in production (current rev.: 4943). For 3 times in 50 days we had to restart openser to correct pkg memory problem. After some time logging messages like this: /openser.log:Aug 19 10:39:18 ipx022 /usr/local/sbin/openser[16991]: ERROR:core:new_credentials: no pkg memory left, openser will eventually run out of pkg memory and refuse all subsequent requests. We are trying to recreate this in our lab so that we can follow memory troubleshooting instructions at http://kamailio.net/dokuwiki/doku.php/troubleshooting:memory, but so far we were unable to do it even when generating millions of calls and registration transactions (we are using SIPp to generate normal call flows and even abnormal call flows detected when reading openser.log, like 'invalid cseq for aor', malformed SIP messages etc). And this is much more than in our production environment, with just 600 subscribers and about 2000 calls a day. The frequency the problem happens is increasing with the number of subscribers, so we are performing periodic restart of openser (actually, what we do is to switch over to the standby server). We already recompiled openser with pkg memory pool size set to 4MB so that this will not have to be done frequently. Since we cannot recreate this in our lab, we suspect there is a situation happening in production that might not be having been properly handled by openser.cfg. So my question is: would it be possible to an overlooked detail in openser.cfg to cause pkg memory problem? In case someone could take a look at it, here's our cfg file: ####### Global Parameters ######### debug=0 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:202.173.5.181:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "carrierroute.so" loadmodule "nathelper.so" loadmodule "dialog.so" loadmodule "snmpstats.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "lcr.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" loadmodule "uac.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 10) modparam("registrar", "min_expires", 30) modparam("registrar", "max_expires", 40) modparam("registrar", "default_expires", 35) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "use_domain", 1) # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "use_domain", 1) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid;$avp(s:blocked)=subscriber_status") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("alias_db", "use_domain", 0) # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") # ----- carrieroute params ----- modparam("carrierroute", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("carrierroute", "config_source", "db") modparam("carrierroute", "use_domain", 1) # ----- NatHelper ----- #para versao a partir da versao 1.2 eh necessario esse paramtro para nao dar erro qdo usa a funcao "fix_nated_register();" modparam("nathelper|registrar", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- LCR ----- modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "dm_flag", 25) # ----- Dialog ---- modparam("dialog", "dlg_flag", 4) # ----- SnmpStat ----- modparam("snmpstats", "sipEntityType", "registrarServer") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 2000) modparam("snmpstats", "MsgQueueMajorThreshold", 5000) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_major_threshold", 750) modparam("snmpstats", "snmpgetPath","/usr/bin/") modparam("snmpstats", "snmpCommunity","public") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } ##nat route(2); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server #t_relay(); #exit; route(1); } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests setflag(4); #for dialog statistics # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) route(1); # t_relay(); # exit; } #t_check_trans(); if (is_method("PUBLISH|SUBSCRIBE|REFER|OPTIONS|MESSAGE")) { sl_send_reply("405", "Method not allowed"); exit; } # authenticate if from local subscriber (uncomment to enable auth) if (!(method=="REGISTER") && (!from_gw())) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; }else if (avp_check("$avp(s:blocked)", "eq/0")) { sl_send_reply("603","Subscriber disabled"); exit; }else if (avp_check("$avp(s:blocked)", "eq/1")) { sl_send_reply("603","Subscriber with outgoing blocked"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; }else if (avp_check("$avp(s:blocked)", "eq/0")) { sl_send_reply("403","Subscriber disabled"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); #if the call came from a known gateway it is not authenticated and we cannot use the function check_from() if (from_gw()) { route(4); }else if (!check_from()) {#if the check_from() returns false the call is not from a subscriber route(4); } else {#it is a subscriber, route using flip domain xlog("L_INFO", "routing using carrierroute $rm to $ru\n"); if (!cr_user_rewrite_uri("$fu", "flip")) { t_newtran(); t_reply("404", "No Route"); exit; } #replaces from by it's default DID uac_replace_from("sip:$avp(s:rpid)@$fd"); } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { xlog("L_INFO", "ROUTE_1 $rm to $ru\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')) { setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { xlog("L_INFO", "ROUTE_2 $rm to $ru\n"); if (method=="REGISTER") { fix_nated_register(); } else if (!from_gw()){ fix_nated_contact(); }; setflag(5); } route[3] { xlog("L_INFO", "ROUTE_3 $rm to $ru\n"); if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { xlog("L_INFO", "FORCE RTP w/ parameter.\n"); force_rtp_proxy("r"); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } route[4] { xlog("L_INFO", "uri does exist $rm to $ru \n"); if (alias_db_lookup("dbaliases")){ if (!lookup("location")) { switch ($retcode) { case -1: t_newtran(); t_reply("404", "Subscriber not online"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } }else{#check if did is blocked $rU = "(BLK)" + $rU; if (alias_db_lookup("dbaliases")){ sl_send_reply("403", "DID blocked"); exit; }else{# if it is not a valid DID nor a blocked DID tries to route it using peering domain if (!cr_rewrite_uri("peering", "call_id")) { t_newtran(); t_reply("404", "Peering Not Found"); exit; } } } } failure_route[1] { xlog("L_INFO", "FAILURE $rm to $ru\n"); if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { xlog("L_INFO", "ONREPLY_1 - Status $rs from $si $rm .\n"); if (is_method("INVITE")) { if ((isflagset(5)||isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (!from_gw()){ #if (isflagset(6)) { xlog("L_INFO", "ONREPLY_1 - ! from gw.\n"); fix_nated_contact(); } exit; } } Regards, takeshi

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Re: [Kamailio-Users] Case Sensitive
by Daniel-Constantin Mierla 07 Oct '08

07 Oct '08

Please cc all the time to mailing lists, so everybody benefits of the outcome or can add comments/suggestions. Also, repeat my question, right now do you need it in some module or for string operations in the config file? Thank you, Daniel On 10/02/08 12:18, michel freiha wrote: > Dear Daniel, > > I prefer doing it in the config file...Is there any function that I > can add and it'll runn the server in case sensitive mode? > > Regards > > On Thu, Oct 2, 2008 at 11:42 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 10/02/08 11:10, michel freiha wrote: > > Dear All, > I would like to ask please about how to force kamailio to be > case sensitive regarding username? > > similar issue reported on the tracker: > http://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=1… > <http://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=1…> > > I guess it is same for you. Right now, do you need it in some > functions or in config file operations? There is need for special > functions for different comparisons, but also might be good to > have case sensitive operations in config file. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com <http://www.asipto.com/> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Dispatcher hash algorithms.
by Alex Balashov 06 Oct '08

06 Oct '08

I am curious, what exactly is the purpose and philosophy of the various hash algorithms in the dispatcher module? I am referring to the ones that allow the gateway in the route set to be determined through a hash of various SIP headers, such as the From URI, the To URI, the Call-ID GUID, etc. No guarantee of fair distribution - or in fact, any distribution that can be characterised by any describable pattern whatsoever - is implied by these algorithms. Additionally, it seems that without knowing (1) The exact hash algorithm in use; (2) The distribution that this hash algorithm would yield for a given set of possible values of these header fields, which in many cases are specifically intended to be pseudorandom (e.g. Call-ID), there is absolutely no way to determine, from any meaningfully deterministic perspective, which numerical entry in the route set these algorithms would computationally yield. So, I guess my question is: With no implied uniformity or weighting in the distribution whatsoever based on the incidental character of such values, what practical use does it serve to use any algorithms except round-robin or random? Is it expected that the user will plot the hash values against a log of given input strings to determine how the distribution will shape up? Is there some reason why the sort of profoundly lopsided distribution that may create might be desirable? Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] CDRTools: Error instantiating db class:
by Tiago Rés 06 Oct '08

06 Oct '08

Hello all, When I login to CDRTool, it appears this error ("Error instantiating db class:") in the main page (CDRs). Any suggestion? Thanks! Tiago

1 0

[Serusers] lcr module and cancel
by Deepak 06 Oct '08

06 Oct '08

Hi, I have the following issue: I am using lcr module to select a gw from a set of gw's based upon R-URI prefixes. All is working fine. The issue is with a CANCEL that is sent by the UAC that originates the call before a dialog can be established. This implies that the CANCEL cannot be routed to the correct GW since SER see a CANCEL and simply returns "487 request terminated". How can I forward the CANCEL to the GW the the original INVITE was forwarded to? Any help appreciated.....

2 4

[Kamailio-Users] [offtopic] update vs delete/insert
by Daniel-Constantin Mierla 06 Oct '08

06 Oct '08

Hello, I am trying to get out you knowledge as DBA ... in some discussions in the past was brought out that in some cases doing a delete on primary key followed by a pure insert might be faster than one update. It might be dependent of DB server type and perhaps the question would have better answer in appropriate db-specific mailing lists, but I want to see if some of you has more experience/insights about this and is willing to share or got into same dilemma. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Carrierroute module: preffix qty?
by Pablo Bertuzzi 06 Oct '08

06 Oct '08

Hi Forum, I want to use carrierroute to terminate 50,000 preffixes to a PSTN gw and block 100,000. So I created 50,000 records with gw ip as rewrite_host. However I only was able to start kamailio with no more than about 2,200 records in carrierroute mysql table. Is it possible to use carrierroute in such a way? Find below log with error: Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:bind_data_loader: use database as configuration source Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:load_route_data: name default, id 0, trees: 1 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: add carrier default Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: carrier default not found, add it Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_tree: tree default has internal id 0 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: created carrier tree: default, with id 0 and 1 trees Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_row: no private memory left Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_rows: error while converting row #2282 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_store_result: error while converting result Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:db_do_query: error while storing result Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:load_route_data: Failed to query database. Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:prepare_route_tree: could not load routing data Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:mod_init: could not prepare route tree Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:init_mod: failed to initialize module carrierroute Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:main: error while initializing modules

3 2

[Kamailio-Users] Developing Sh interface between the AS(OPENSER/OPENSIPS)and HSS
by suresh tattikota 06 Oct '08

06 Oct '08

hi all I am using the Openser(Opensips) as apllication server . i am developing Sh interface between the AS and HSS. For developing the Sh interface i am using the CDiameterpeer , here i am getiing memory and locking errors. And for using the CDiameterpeer as standalone library what steps i have to follow ... Thanks suresh :)

2 1

Re: [Kamailio-Users] nested redirect
by Daniel-Constantin Mierla 05 Oct '08

05 Oct '08

Hello, your main route block is something loke: route { if(is_method("INVITE")) { .... } exit; } So, apart of INVITE, every other sip request is dropped. See default config file to get a better understanding. kamailio/openser discards by default any sip request received if there is no action to forward for it. Cheers, Daniel Kent Pirlo wrote: > Attached is the full test config I am using.. I have gone back and forth trying a few things, but this is the latest.. If you have a another way to achieve the same result, I am open to change how I'm trying to do it. > > Thanks again for your help. > > Kent > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, September 23, 2008 5:50 AM > To: Kent Pirlo > Cc: users > Subject: Re: [Kamailio-Users] nested redirect > > Hello, > > sorry, you were right, there was a pcap attachment! The ACK looks ok. > Seems that ACK is quickly discarded by the script. Is the config you > pasted in your first email complete? I need to see it all the understand > how the ack get's routed and where could be dropped by the config. > > Cheers, > Daniel > > > > On 09/17/08 18:42, Kent Pirlo wrote: > >> There should have been a 20k attachment.. redirect.cap.. from tethereal. >> Let me know if you do not have it, and I will re-send it. >> >> kent >> >> -----Original Message----- >> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Sent: Wednesday, September 17, 2008 6:43 AM >> To: Kent Pirlo >> Cc: users >> Subject: Re: [Kamailio-Users] nested redirect >> >> Hello, >> >> what you sent is the debug from kamailio/openser, which is useful, but I >> would need the sip messages, you can use ngrep for that, do >> ngrep -d any -qt -W byline port 5060 >> >> on your openser/kamailio server and send the output. >> >> Thanks, >> Daniel >> >> >> On 09/17/08 02:22, Kent Pirlo wrote: >> >> >>> ... is my core switch, >>> ... is my openser/Kamailio redirect server >>> ... is a redirect server from a carrier of ours that returns a billing TN in contact header.. >>> >>> >>> Here is the debug of the incoming ACK from the core switch after being sent the 300... >>> >>> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: SIP Request: >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: method: <ACK> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: uri: <sip:...@...> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: version: <SIP/2.0> >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: flags=2 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to_param: tag=391854ebdd4d0c63cf50836cbb12191e-6708 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to: end of header reached, state=29 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to: display={}, >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: <To> [75]; >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: to body >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: cseq <CSeq>: <1> <ACK> >>> Sep 16 15:49:50 [6505] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKdce3ac34730f843a5a9b9c7002690f22>; state=16 >>> Sep 16 15:49:50 [6505] DBG:core:parse_via: end of header reached, state=5 >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: via found, flags=2 >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: this is the first via >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: After parse_msg... >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: preparing to run routing scripts... >>> Sep 16 15:49:50 [6505] DBG:sl:sl_filter_ACK: to late to be a local ACK! >>> Sep 16 15:49:50 [6505] DBG:core:destroy_avp_list: destroying list (nil) >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: cleaning up >>> >>> >>> >>> >>> -----Original Message----- >>> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Sent: Tuesday, September 16, 2008 5:10 AM >>> To: kentp(a)globalpops.com >>> Cc: users(a)lists.kamailio.org >>> Subject: Re: [Kamailio-Users] nested redirect >>> >>> Hello, >>> >>> can you post here the SIP messages? Will help to spot if there is >>> anything wrong with the ACK or something else in the signaling. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 09/13/08 04:11, kentp(a)globalpops.com wrote: >>> >>> >>> >>>> i have a simple redirect server working: >>>> >>>> Call comes from switch (A) to kamailio redirect server (B).. >>>> (B) runs perl script to determine route >>>> (B) sends Stateless 300 redirect reply to (A). >>>> (A) sends ACK to (B). >>>> >>>> Now, I need to add a nested redirect... >>>> >>>> call comes from sitch (A) to kamailio redirect server (B).. >>>> (B) then needs to send the call to third party proxy (C).. >>>> (C) will return 302 redirect.. >>>> (B) sends ACK to (C) >>>> (B) needs to read the contact header, then run perl script to determine route (using the contact value >>>> returned from (c)) >>>> (B) then sends 300 redirect back to (A). >>>> (A) sends ACK back to (B) >>>> >>>> >>>> so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending >>>> multiple 300's to (A): >>>> >>>> route { >>>> route(1); >>>> } >>>> >>>> route[1] { >>>> t_on_reply("1"); >>>> t_on_failure("2"); >>>> if (!t_relay("proxy.server.c:5060","0x02")) { >>>> sl_reply_error(); >>>> } >>>> onreply_route[1] { >>>> if(t_check_status("302")){ >>>> $var(lrn_from_bv) = $ct; >>>> } >>>> } >>>> failure_route[2] { >>>> >>>> if (t_check_status("3[0-9][0-9]")) { >>>> route(2); >>>> exit; >>>> } >>>> route[2] { >>>> if(perl_exec("callcheck")){ >>>> avp_pushto("$ru", "$avp(s:routes)/g"); >>>> t_reply("300","Go Here"); >>>> exit; >>>> } >>>> } >>>> >>>> >>>> >>>> -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] LCR module and CANCEL
by Deepak 04 Oct '08

04 Oct '08

Hi, I have the following issue: I am using lcr module to select a gw from a set of gw's based upon R-URI prefixes. All is working fine. The issue is with a CANCEL that is sent by the UAC that originates the call before a dialog can be established. This implies that the CANCEL cannot be routed to the correct GW since SER see a CANCEL and simply returns "487 request terminated". How can I forward the CANCEL to the GW the the original INVITE was forwarded to? Any help appreciated.....

1 0

[Kamailio-Users] Presence Server returns a 404 Not here msg
by Yoo Chan Jeon 03 Oct '08

03 Oct '08

Hi, I setup the presence server. My sip phone sends a PUBLISH msg. Th presence server sends a 200 msg. My another sip phone sends a SUBSCRIBE msg. The presence server returns 202 msg, and NOTIFY msg. These msgs does not include user in the contact header My sip phone sends a 200 msg I sends a second SUBSCRIBE msg to refresh it. This time my subscribe msg does not include the user in the request msg because the presence server did not include the user in the contact header. The presence server replied back with a 404 messge. So I looked at the codes, and add the user in the contact header when the presence server replys back to a client. But I stll have a same 404 msg. Do I miss something? I always appreciate your help. Thanks. Jeon

2 1

[Kamailio-Users] openser.cfg children=4
by Asim Riaz 03 Oct '08

03 Oct '08

Hi, my openser.cfg is configured for childern=4, but I am seeing child processes? Thanks Asim Riaz

3 6

Re: [Kamailio-Users] sermyadmin setup suse 11
by Ricky Gutierrez 03 Oct '08

03 Oct '08

I have looked for this inside the kamctl and I don't find it, I don't believe that it was that the export error, somebody that has installed sermyadmin in suse 11? regards .. rickygm Wednesday, October 1, 2008 12:24:06 PM, Iñaki Baz Castillo wrote: Subject: Re: [Kamailio-Users] sermyadmin setup suse 11 Please, don't send an image attachment, just copy the console output. This error is very common and is related to "phplib" field. Modify /etc/openserctl and set it to "no" (or something similar..). Look for the exact error in Google for more responses. --

2 1

[Kamailio-Users] Case Sensitive
by michel freiha 02 Oct '08

02 Oct '08

Dear All, I would like to ask please about how to force kamailio to be case sensitive regarding username? Regards

2 1

[Kamailio-Users] registrar module enhancements
by Daniel-Constantin Mierla 02 Oct '08

02 Oct '08

Hello, in the last time several enhancements were added to registrar module to help in dealing with user location records. This correlated with the new modue pv shall give lot of control over the destination set for calls. To get the latest devel version, check out the latest svn - svn co https://openser.svn.sourceforge.net/svnroot/openser/trunk kamailio The news in registrar module are: - ability to unregister all contacts for a SIP addres - there is a new function: unregister(table, uri) - ability to store and maintain single contact per user - the save() function accepts a new value for flags parameter (0x04) to restrict to one single record in location table - ability to access the attributes of all contacts for a user - there are two new functions: reg_fetch_contacts(table, uri, profile) and reg_free_contacts(profile) to make available/release the contacts for a SIP uri. After fetching the contacts, the attributes can be accessed via pseudo variable $ulc(profile=>attr) For more details see the README: http://www.kamailio.org/docs/modules/devel/registrar.html Here are couple of use cases: 1) only one contact per user save("location", "0x04"); Note that if you have couple of phones registering for same user, the one that sent last REGISTER message will be the only ringing. 2) check if a user is calling from a registered device and if not, deny the call if(!reg_fetch_contacts("location", "$fu", "caller")) { sl_send_reply("403", "Please register first"); exit; } $var(i) = 0; $var(found) = 0; $var(contact) = $(ct{nameaddr.uri}); while($var(found) == 0 && $var(i) < $(ulc(caller=>count))) { $if($var(contact)==$(ulc(caller=>addr)[$var(i)])) $var(found) = 1; else $var(i) = $var(i) + 1; } if($var(found) == 0) { sl_send_reply("403", "Please register first"); exit; } Testing and feedback is very much appreciated. For the future: - unregister() to get a $ulc(...) variable as parameter to unregister a specific contact - drop a branch via pv module Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 5

[Kamailio-Users] RLS(Resource List Server)
by Yoo Chan Jeon 02 Oct '08

02 Oct '08

Hi, I just want to know how RLS handles SUBSCRIBE messages without using XCAP. I assume that the server already has the list information. I want send the SUBSRIBE message with resource list format: SUBSCRIBE sip:list@domain . . supported:eventlist Accept:application/rlmi+xml Accept:multipart/related I added rls, presence, presence_xml, and pua modlues in the configuration file. I do not have any problems to run the server. Which database should I add list@doamin <list@doamin> to? How do I add the resource list? What kind of format do I need to add a list if I want to add joe@domain, bob@domain, tom@domain ? Do you have some kind of examples? Thanks you. Jeon

2 1

[Kamailio-Users] sermyadmin setup suse 11
by Ricky Gutierrez 01 Oct '08

01 Oct '08

Hi, I am trying to install sermyadmin 0.8 with the patch of security, but when I export the file openser.sql, it shows me this error: INSERT INTO `jsec_user_role_rel` (`id`, `version`, `role_id`, `user_id`) V' at line 1 if I try to export it again, it shows me an error message of having duplicated domain I erase the data of the domain chart, but same the same error ERROR 1062 (23000) at line 6: Duplicate entry '1' for key 1 God blesses, any help is welcome rickygm

2 1

[Serusers] two way routing
by Ivan Zweig 01 Oct '08

01 Oct '08

Hi, we are a small hosted VOIP company. We need a box that can take IN multiple origination providers and route the calls to the appropriate customer instance on our hosted platform. We use switchvox. We would also like to have 2-3 outbound routes to gain price efficiency and redundancy. I need someone that can implement this box and train us how to use it. you can email me to start a dialogue. We are looking for a consultant to help us with this.

1 0

[Kamailio-Users] About presence_dialoginfo - RR in figure 2
by Iñaki Baz Castillo 01 Oct '08

01 Oct '08

Hi, figure 2 of http://www.kamailio.org/docs/modules/devel/presence_dialoginfo.html shows the presence server sending the NOTIFY directly to the watcher. Is it possible to send the NOTIFY through the proxy? Well, since NOTIFY is in-dialog it's just needed that the proxy does RR in the initial SUBSCRIBE before forwarding it to presence server, am I right? Thanks. -- Iñaki Baz Castillo

3 3

[Kamailio-Users] new CDRTool release 6.6.9
by Adrian Georgescu 01 Oct '08

01 Oct '08

Hello, The software contains bug fixes for parallel prepaid calls. Changelog since 6.6.8 cdrtool (6.6.9) unstable; urgency=low * Purge old entries from prepaid_history table * Fixed wrong return to call control engine because sessionDoesNotExist variable was not initialized properly at each request * Print the call id in the syslog when calculating rate The software can be downloaded from: http://download.ag-projects.com/CDRTool/ For people running Debian testing or unstable on an i386 architecture there is an official public repository provided by AG Projects. To use it, add these lines in /etc/apt/sources.list # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main Install the AG Projects debian software signing key: wget http://download.ag-projects.com/agp-debian-gpg.key apt-key add agp-debian-gpg.key After that, run: apt-get update apt-get install cdrtool Regards, Adrian

1 0

[Kamailio-Users] LCR vs Carrierroute and SRV Support Inquirer
by Krom 30 Sep '08

30 Sep '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for RURI manipulation prior to relaying to a cluster of session boarder controllers. As you know the DB table for LCR only supports integers in the gw table for ip_addr. I have been reading up on 1.2.x and 1.3.x to upgrade to a solution that would allow us to stay within the same call processing (script) design, but still support the new upstream requirement. I noticed the carrierroute table rewrite_host column is string. Could this be used to satisfy my needs of manipulating the RURI for an IP where the carrier still uses a list of gateway IP's to try and use the DNS/SRV domain name where the carrier requires the use for SRV look-up? The session boarder controllers will conduct the SRV look-up after receiving the relayed message from Kamailio, it is just getting the domain name on the RURI instead of IP address that I am looking to accomplish. Will the (new to me) carrierroute module assist in this effort or is there another way that I am totally missing? Figured I would ask the gurus before following Alice down the rabbit hole. -- Thanks, -- Kyle

1 0

[Kamailio-Users] How to prevent - Same username used on multiple sip phones
by Ali Jawad 30 Sep '08

30 Sep '08

Hi On my openser I can logon with the same username on two different softphones. How can I prevent the same username from being used on multiple devices. With Regards

6 12

[Serusers] how-to AVP from radius load
by Samuel Muller 30 Sep '08

30 Sep '08

Hello, I've a single question about AVPs : how can I use them when loaded from radius_load_attr ? example : the 0123452000 is the caller, I want to use an acl to block him. => in my radius user config : 0123451010(a)sip.dot.com User-Password = "0000" User-Name = "0123451010(a)sip.dot.com", SER-UID = "0123451010(a)sip.dot.com", SER-DID = "1", SER-From-UID = "0123451010(a)sip.dot.com", SER-From-DID = "1", SER-Digest-Username = "0123451010", SER-Digest-Realm = "sip.dot.com", SER-Uri-User = "0123451010(a)sip.dot.com", SER-Attr = "acl:pstn", SER-Attr = "asserted-id:0123451010", SER-Attr = "phone:0123451010", SER-Attr = "lang:fr" => in the SER (v2.0.0) : if (!radius_load_attrs("$fu", "@f.uri")) { sl_reply("403", "Radius attributes not found"); exit; } if ( $acl == "pstn" ) { sl_reply("401", "Unauthorized"); exit; } I tried too " $fu.acl ", " set_attr function, ... it seems I can't use SER-Attr as a variable. in the debug (dump_attrs) : ... ser[28670]: r_INBOUND: => dump_attrs() ser[28670]: INFO: avp.c:542: class=GLOBAL ser[28670]: AVP["lang"]="en" ser[28670]: INFO: avp.c:552: track=FROM class=DOMAIN ser[28670]: AVP["did"]="1" ser[28670]: AVP["digest_realm"]="sip.dot.com" ser[28670]: INFO: avp.c:562: track=TO class=DOMAIN ser[28670]: AVP["did"]="1" ser[28670]: AVP["digest_realm"]="sip.dot.com" ser[28670]: INFO: avp.c:572: track=FROM class=USER ser[28670]: AVP["uid"]="0123451010(a)sip.dot.com" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451010" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451010" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451010" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451010" ser[28670]: INFO: avp.c:582: track=TO class=USER ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451005" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451005" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451005" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451005" ser[28670]: INFO: avp.c:592: track=FROM class=URI ser[28670]: AVP["uac_nat"]=1 ser[28670]: AVP["account"]="yes" ser[28670]: INFO: avp.c:602: track=TO class=URI ser[28670]: INFO: No AVP present ... If anyone has tips to help me, many thanks ! I'm sure there's something who replaces the old group_radius module, but which one or how to create SER-Attr $grp ? Many thanks, -- Samuel MULLER sml(a)720.fr

1 0

[Kamailio-Users] new CDRTool release 6.6.8
by Adrian Georgescu 30 Sep '08

30 Sep '08

Hello, The software includes critical bug fixes for parallel prepaid calls. The software can be downloaded from: http://download.ag-projects.com/CDRTool/ Changelog since 6.6.3 cdrtool (6.6.8) unstable; urgency=low * Log debit balance to prepaid_history table cdrtool (6.6.7) unstable; urgency=low * Do not debit balance for inexistent prepaid sessions cdrtool (6.6.5) unstable; urgency=low * Fixed purging of expired prepaid sessions cdrtool (6.6.4) unstable; urgency=low * Added links to Prepaid calls and quota in the main menu * Fixed showFooter() when exporting csv files from rating page * Fixed prepaid table mysql schema Regards, Adrian

1 0

[Kamailio-Users] kamailio 1.4.1 and mediaproxy 1.9.1
by toly 29 Sep '08

29 Sep '08

Greetings, I'm looking for the short answer, most likely no way. Thanks, Toly -- View this message in context: http://www.nabble.com/kamailio-1.4.1-and-mediaproxy-1.9.1-tp19728623p197286… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] Disable transfers (REFER) on Kamailio
by Diego Zuaneti Arruda 29 Sep '08

29 Sep '08

Hi all, It's possible to disable any transfers (blind or attended) on Kamailio ? Thanks. Diego.

2 2

Re: [Kamailio-Users] SDP ending with CRLF and a=nortpproxy:yes
by Schober Walter 29 Sep '08

29 Sep '08

I see: * [r4968] modules/nathelper/nathelper.c: - fix the calculation of CRLF length when adding norttptoxy string - patch by Alex Hermann (closes #2117700) And yes, it's that issue and therefore fixed in 1.4.1. > -----Ursprüngliche Nachricht----- > Von: Schober Walter > Gesendet: Samstag, 27. September 2008 14:13 > An: 'miconda(a)gmail.com'; Jerome Martin > Cc: users(a)lists.kamailio.org > Betreff: AW: [Kamailio-Users] SDP ending with CRLF and > a=nortpproxy:yes > > Hi! > > Same issue with 1.4.0. I'm migrating from 1.1.1 to 1.4.0, but > force_rtp_proxy() now adds a sdp line with a beginning 0x00, > so line looks like > 0000 00 61 3d 6e 6f 72 74 70 70 72 6f 78 79 3a 79 65 > .a=nortpproxy:ye > 0010 73 s > But no \r\n\r\n in front anymore in 1.4.0. > > Someone to direct me to the propper source file,pls? > > Walter > > > -----Ursprüngliche Nachricht----- > > Von: users-bounces(a)lists.kamailio.org > > [mailto:users-bounces@lists.kamailio.org] Im Auftrag von > > Daniel-Constantin Mierla > > Gesendet: Montag, 15. September 2008 14:23 > > An: Jerome Martin > > Cc: users(a)lists.kamailio.org > > Betreff: Re: [Kamailio-Users] SDP ending with CRLF and > > a=nortpproxy:yes > > > > Hi Jerome, > > > > On 09/15/08 13:43, Jerome Martin wrote: > > > Hi All, > > > > > > Just FYI, I had to disable the nortpproxy test here because > > it breaks > > > thomson ST2030 endpoints : with that line, they would just plain > > > ignore the messages containing the SDP ... > > was i with malformed SDP (empty lines inside SDP before the > a= line), > > or just the presence of the a=nortpproxy:yes line made the > ST2030 go > > mad? > > > > Cheers, > > Daniel > > > > > > > > Cheers, > > > > > > On Fri, 2008-09-12 at 18:12 +0300, Daniel-Constantin Mierla wrote: > > >> On 09/11/08 16:10, Klaus Darilion wrote: > > >> > A similar bug was also described in opensips. They tried > > to fix it > > >> > but the bug is still present. I wonder if this is > > something new or > > >> > if the bug is thre since the beginning of nathelper > but did not > > >> > occured until now > > >> > > > >> maybe this can be fixed by using the sdp parser available > > now in core. > > >> nathelper has quite limited sdp parsing capabilities now. > > the issue > > >> is to find the proper hook to add the lump with > > >> > > >> a=nortpproxy:yes\r\n" > > >> > > >> Cheers, > > >> Daniel > > >> > > >> > > >> > klaus > > >> > > > >> > Aymeric Moizard schrieb: > > >> > > > >> >> Hi, > > >> >> > > >> >> I just happen to find an issue with a software trying to call > > >> >> through my openser-1.3.1 > > >> >> > > >> >> After inserting "a=nortpproxy:yes\r\n", the message > coming out > > >> >> openser is: > > >> >> > > >> >> [normal sdp packet] > > >> >> a=fmtp:101 0-15\r\n > > >> >> \r\n > > >> >> \r\n > > >> >> "a=nortpproxy:yes\r\n" > > >> >> > > >> >> No matter who is responsible in that specific case (the SDP > > >> >> received by openser with several \r\n looks not > > compliant to me). > > >> >> > > >> >> I think openser should add "a=nortpproxy:yes" in > > different place > > >> >> in the SDP packet: NOT at the end of the SDP body. The > > "a=nortpproxy:yes" > > >> >> should be either put at the global attribute level or > > right below > > >> >> a m= line. > > >> >> > > >> >> This approach would solve such issue and would be much more > > >> >> adequate to the requirement: currently it's not very > > possible to > > >> >> relay an audio stream and not relay a video stream with the > > >> >> current approach. > > >> >> > > >> >> tks, > > >> >> Aymeric MOIZARD / ANTISIP > > >> >> amsip - http://www.antisip.com > > >> >> osip2 - http://www.osip.org > > >> >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > >> >> > > >> >> > > >> >> _______________________________________________ > > >> >> Users mailing list > > >> >> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > > >> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > >> >> > > >> > > > >> > _______________________________________________ > > >> > Users mailing list > > >> > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > > >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > >> > > > >> > > > >> > > >> > > > > > > *Jérôme Martin **| **LongPhone* > > > *Responsable Architecture Réseau* > > > 122, rue la Boetie | 75008 Paris > > > Tel : +33 (0)1 56 26 28 44 > > > Fax : +33 (0)1 56 26 28 45 > > > Mail : *jmartin*(a)longphone.fr > > > Web : www.longphone.com <http://www.longphone.com> > > > > > > > > > > > > ---------------------------------------------------------------------- > > > -- > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Daniel-Constantin Mierla > > http://www.asipto.com > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] CDRTool - stored procedures - not creating tables
by Graham Wooden 28 Sep '08

28 Sep '08

All, I have been having a heck of a time getting the radacct tables to be generated from the stored procedures, with the latest CDRTool. I have done and have reviewed them again the steps mentioned at: http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt Can anyone give any guidance. The call entries are showing up (for the most part) in the radacct table, but getting the main error upon viewing the CDR. ³Database error: Invalid SQL: select count(*) as records from radacct200809 where (AcctStartTime >= '2008-09-28 05:34' and AcctStartTime < '2008-09-28 23:55') MySQL error: 1146 (Table 'cdrtool.radacct200809' doesn't exist) 64Session halted.² I am using 1.4 Kamailio Thanks, -graham

1 0

Re: [Kamailio-Users] SDP ending with CRLF and a=nortpproxy:yes
by Schober Walter 27 Sep '08

27 Sep '08

Hi! Same issue with 1.4.0. I'm migrating from 1.1.1 to 1.4.0, but force_rtp_proxy() now adds a sdp line with a beginning 0x00, so line looks like 0000 00 61 3d 6e 6f 72 74 70 70 72 6f 78 79 3a 79 65 .a=nortpproxy:ye 0010 73 s But no \r\n\r\n in front anymore in 1.4.0. Someone to direct me to the propper source file,pls? Walter > -----Ursprüngliche Nachricht----- > Von: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] Im Auftrag von > Daniel-Constantin Mierla > Gesendet: Montag, 15. September 2008 14:23 > An: Jerome Martin > Cc: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] SDP ending with CRLF and > a=nortpproxy:yes > > Hi Jerome, > > On 09/15/08 13:43, Jerome Martin wrote: > > Hi All, > > > > Just FYI, I had to disable the nortpproxy test here because > it breaks > > thomson ST2030 endpoints : with that line, they would just plain > > ignore the messages containing the SDP ... > was i with malformed SDP (empty lines inside SDP before the > a= line), or just the presence of the a=nortpproxy:yes line > made the ST2030 go mad? > > Cheers, > Daniel > > > > > Cheers, > > > > On Fri, 2008-09-12 at 18:12 +0300, Daniel-Constantin Mierla wrote: > >> On 09/11/08 16:10, Klaus Darilion wrote: > >> > A similar bug was also described in opensips. They tried > to fix it > >> > but the bug is still present. I wonder if this is > something new or > >> > if the bug is thre since the beginning of nathelper but did not > >> > occured until now > >> > > >> maybe this can be fixed by using the sdp parser available > now in core. > >> nathelper has quite limited sdp parsing capabilities now. > the issue > >> is to find the proper hook to add the lump with > >> > >> a=nortpproxy:yes\r\n" > >> > >> Cheers, > >> Daniel > >> > >> > >> > klaus > >> > > >> > Aymeric Moizard schrieb: > >> > > >> >> Hi, > >> >> > >> >> I just happen to find an issue with a software trying to call > >> >> through my openser-1.3.1 > >> >> > >> >> After inserting "a=nortpproxy:yes\r\n", the message coming out > >> >> openser is: > >> >> > >> >> [normal sdp packet] > >> >> a=fmtp:101 0-15\r\n > >> >> \r\n > >> >> \r\n > >> >> "a=nortpproxy:yes\r\n" > >> >> > >> >> No matter who is responsible in that specific case (the SDP > >> >> received by openser with several \r\n looks not > compliant to me). > >> >> > >> >> I think openser should add "a=nortpproxy:yes" in > different place > >> >> in the SDP packet: NOT at the end of the SDP body. The > "a=nortpproxy:yes" > >> >> should be either put at the global attribute level or > right below > >> >> a m= line. > >> >> > >> >> This approach would solve such issue and would be much more > >> >> adequate to the requirement: currently it's not very > possible to > >> >> relay an audio stream and not relay a video stream with the > >> >> current approach. > >> >> > >> >> tks, > >> >> Aymeric MOIZARD / ANTISIP > >> >> amsip - http://www.antisip.com > >> >> osip2 - http://www.osip.org > >> >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ > >> >> > >> >> > >> >> _______________________________________________ > >> >> Users mailing list > >> >> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > >> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> >> > >> > > >> > _______________________________________________ > >> > Users mailing list > >> > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > >> > > >> > >> > > > > *Jérôme Martin **| **LongPhone* > > *Responsable Architecture Réseau* > > 122, rue la Boetie | 75008 Paris > > Tel : +33 (0)1 56 26 28 44 > > Fax : +33 (0)1 56 26 28 45 > > Mail : *jmartin*(a)longphone.fr > > Web : www.longphone.com <http://www.longphone.com> > > > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] towards kamailio v1.5.0
by Daniel-Constantin Mierla 26 Sep '08

26 Sep '08

Hello, as we just released 1.4.1 in the latest stable branch, I just want to give updates on the development version of the project. In less that two months since 1.4.x release series was branched, lot of new things happened - new modules and features, code refactoring and fixes. The outline of the new additions: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x A drafted version of the roadmap is available at: http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x I take this occasion to ask all developers to add there the new features already introduced and their plans for 1.5.0 in the above two pages. There are some features introduced on SVN that are not yet listed in Dokuwiki, features:new-in-1.5.x. So please, when you add some major improvement, add a short note about. As shortcut, those links are hosted on the website page: http://www.kamailio.org/mos/view/Roadmap/ For a deeper track of all changes you can browse the SVN repository at Source Forge. http://openser.svn.sourceforge.net/viewvc/openser/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] ip address of VIA header
by Alex R.S.M 26 Sep '08

26 Sep '08

Hi, Is there any function in openSER to return the ip address of VIA header? or return the VIA header? Thanks, alex _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

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[Kamailio-Users] Exported statistics
by Daniel Cardoso Alves 26 Sep '08

26 Sep '08

Hi people... I am trying to use "Exported statistics" of dialog module, but I have the following message: 404 Statistics Not Found I am using the dialog module to finish some calls... And it's work perfect... But I want to get some statistics... I looked on Internet but I dont find a solution... Are there some special configuration to make to use "Exported statistics"??? Thanks in advance... Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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[Kamailio-Users] SUBSCRIBE message with a private header
by Yoo Chan Jeon 26 Sep '08

26 Sep '08

Hi, Thank you for your reponse. By using $hdr(HEADER-NAME), it returns "Private-Header:value". Now I have other problems. I would like to check if the header exist, then I want to get the value. How do I get the value from $hdr(HEADER-NAME) ? Thanks. Jeon El Miércoles, 24 de Septiembre de 2008, Yoo Chan Jeon escribió: >* Hi, *>* I have a client sip agent which sends a SUBSCRIBE msg with a private header *>* to the Kamailio presence server. *>* Hwo do I get the private header informtaion? I want to read it and send it *>* with a Notify message. * Read it with $hdr(HEADER-NAME).

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Re: [Kamailio-Users] How to prevent - Same username used on multiple sip phones
by Ali Jawad 26 Sep '08

26 Sep '08

Hi I did try if(is_method("REGISTER")) { if(lookup("location")){ sl_send_reply("401","Unautorized"); exit; } } With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554 ________________________________ From: Sergio Gutierrez [mailto:saguti@gmail.com] Sent: 2008-09-26 14:38 To: Ali Jawad Subject: Re: [Kamailio-Users] How to prevent - Same username used on multiple sip phones Hi Ali. Maybe you could use the function registered() of the module registrar, to check whether the user is already registered. http://kamailio.org/docs/modules/1.4.x/registrar.html#id2506815 Best regards. Sergio G. On 9/26/08, Ali Jawad <ali.jawad(a)splendor.net> wrote: Hi On my openser I can logon with the same username on two different softphones. How can I prevent the same username from being used on multiple devices. With Regards _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean.

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[Kamailio-Users] SUBSCRIBE message with a private header
by Yoo Chan Jeon 26 Sep '08

26 Sep '08

Hi, Now I got the value. Discard the previous message. Thanks. Jeon >Hi, >Thank you for your reponse. >By using $hdr(HEADER-NAME), it returns "Private-Header:value". >Now I have other problems >I would like to check if the header exist, then I want to get the value. >How do I get the value from $hdr(HEADER-NAME) ? >Thanks. >Jeon El Miércoles, 24 de Septiembre de 2008, Yoo Chan Jeon escribió: > Hi, > I have a client sip agent which sends a SUBSCRIBE msg with a private header > to the Kamailio presence server. > Hwo do I get the private header informtaion? I want to read it and send it > with a Notify message. Read it with $hdr(HEADER-NAME).

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[Kamailio-Users] Failed gateway recovery with dispatcher.
by Alex Balashov 25 Sep '08

25 Sep '08

How does one go about resurrecting gateways that have been marked into "probing" mode back into the "active" set once they start replying to the pings that the dispatcher facility sends? I see the OPTIONS pings going through, and I see the UAs generating replies to them whey come back up. But OpenSER does nothing in response to receipt of those replies to mark them as active again. Because the OPTIONS request is not originated from script, there is no way to catch the replies in an ONREPLY-ROUTE, otherwise I would have tried to handle it that way and re-mark the gateways. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Kamailio v1.4.1 Released
by Daniel-Constantin Mierla 25 Sep '08

25 Sep '08

Hello, A new minor release is available as Kamailio 1.4.1. It is based on the latest version of branch 1.4, therefore those running 1.4.0 are advised to upgrade. The compatibility of configuration file and database structure with v1.4.0 is preserved. This release includes fixes and improvements to issues discovered since the release of v1.4.0. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.1/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.1/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.4.1/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ <http://www.kamailio.org/docs/modules/1.4.x/> Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] crazy error OpenSER 1.3.3
by VIP Carrier 25 Sep '08

25 Sep '08

Hello, I'm trying to compile OpenSER 1.3.3 and getting the following error i have run a comand "make all include_modules="mysql"" -------------------------------------------------------------------------------------------------------- dbase.c:29:25: error: mysql/mysql.h: No such file or directory dbase.c:30:26: error: mysql/errmsg.h: No such file or directory dbase.c:31:33: error: mysql/mysql_version.h: No such file or directory In file included from dbase.c:38: my_con.h:38: error: expected specifier-qualifier-list before Б─≤MYSQL_RESБ─≥ dbase.c: In function Б─≤db_mysql_submit_queryБ─≥: dbase.c:65: error: Б─≤struct my_conБ─≥ has no member named Б─≤timestampБ─≥ dbase.c:66: warning: implicit declaration of function Б─≤mysql_pingБ─≥ dbase.c:66: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:70: error: Б─≤struct my_conБ─≥ has no member named Б─≤timestampБ─≥ dbase.c:88: warning: implicit declaration of function Б─≤mysql_queryБ─≥ dbase.c:88: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:91: warning: implicit declaration of function Б─≤mysql_errnoБ─≥ dbase.c:91: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:92: error: Б─≤CR_SERVER_GONE_ERRORБ─≥ undeclared (first use in this function) dbase.c:92: error: (Each undeclared identifier is reported only once dbase.c:92: error: for each function it appears in.) dbase.c:92: error: Б─≤CR_SERVER_LOSTБ─≥ undeclared (first use in this function) dbase.c:96: warning: implicit declaration of function Б─≤mysql_errorБ─≥ dbase.c:96: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:96: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:96: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c: In function Б─≤db_mysql_store_resultБ─≥: dbase.c:195: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:195: warning: implicit declaration of function Б─≤mysql_store_resultБ─≥ dbase.c:195: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:196: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:197: warning: implicit declaration of function Б─≤mysql_field_countБ─≥ dbase.c:197: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:202: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:202: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:202: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c:216: warning: implicit declaration of function Б─≤mysql_free_resultБ─≥ dbase.c:216: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:223: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c: In function Б─≤db_mysql_free_resultБ─≥: dbase.c:253: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:254: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c: In function Б─≤db_mysql_fetch_resultБ─≥: dbase.c:361: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:361: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:362: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:363: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:368: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:368: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:368: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c:379: warning: implicit declaration of function Б─≤mysql_num_rowsБ─≥ dbase.c:379: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:414: error: Б─≤struct my_conБ─≥ has no member named Б─≤rowБ─≥ dbase.c:414: warning: implicit declaration of function Б─≤mysql_fetch_rowБ─≥ dbase.c:414: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:415: error: Б─≤struct my_conБ─≥ has no member named Б─≤rowБ─≥ dbase.c:416: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:416: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:416: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c: In function Б─≤db_last_inserted_idБ─≥: dbase.c:649: warning: implicit declaration of function Б─≤mysql_insert_idБ─≥ dbase.c:649: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ make[1]: *** [dbase.o] Error 1 make[1]: Leaving directory `/src/openser-1.3.3-notls/modules/mysql' make: *** [modules] Error 2

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[Kamailio-Users] Res: Problems with SerMyAdmin Installation
by Flavio Goncalves 25 Sep '08

25 Sep '08

Hi Brian, I have seen some people complianing about this, but I have not being able to reproduce the problem. Could you send me a copy of your database schema, it is probably the source of the problem. Have you applied the openser.sql in the latest version? Best regards, Flavio E. Goncalves ----- Mensagem original ---- De: Brian Chamberlain <brian(a)asterisk.ie> Para: users(a)lists.kamailio.org Enviadas: Quarta-feira, 10 de Setembro de 2008 13:45:04 Assunto: Re: [Kamailio-Users] Problems with SerMyAdmin Installation Hi, Can you add your users to groups? I just went back to openser 1.2 because of the problems I was facing with sermyadmin adding users to groups and bombing out. I was using 1.33 - but it seems that that is not the problem. I can add groups ok but get the following when adding users to those groups! (GRRRR) Grails Runtime Exception Error Details Message: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] Class: UserController At Line: [115] Code Snippet: Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at UserController$_closure6_closure13.doCall(UserController.groovy:115) at UserController$_closure6.doCall(UserController.groovy:105) at UserController$_closure6.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at $Proxy3.saveOrUpdate(Unknown Source) ... 32 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry 'brian-sip.asterisk.ie-' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 33 more Hello, I solved my problem with Nuno's help. My problem was with the permissions in mysql. The command GRANT ALL PRIVILEGES ON openser.* TO user@'%' IDENTIFIED BY 'Password'; did't add the user properly.. I had to also use GRANT ALL PRIVILEGES ON openser.* TO user@'localhost' IDENTIFIED BY 'password'; because sermyadmin and mysql were running in the same machine. After that I could run sermyadmin and it changed all the tables properly. -- View this message in context: http://www.nabble.com/Problems-with-SerMyAdmin-Installation-tp16965040p1941… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] problem with emergency calls (short-numbers)
by BERGANZ François 25 Sep '08

25 Sep '08

Hello, I have a problem with my emergency calls if (uri=~"^sip:1[578]{1}@*" || uri=~"^sip:11[29]{1}@*") if I call 170725014, it think that it is 17 (emergency in france) is my "if" syntaxe is correct ? Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

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[Kamailio-Users] BAD URI After $ru and avp
by Ali Jawad 25 Sep '08

25 Sep '08

Hi I am trying to change the Ruri as stated below DB usr_preferences Domain= Value=callfwd Attribute=33179974674 if(avp_db_load("$ruri/username","$avp(s:callfwd)")){ xlog("L_INFO","X0XOXOXOXOXOXOXXOXOXOXXOXOXOXOXOX"); $ru = $avp(s:callfwd); route(8); exit; }; It enters allright and it changes the rURI however It throws the errors below: Sep 25 13:31:23 [24879] DBG:avpops:db_close_query: close avp query Sep 25 13:31:23 [24879] DBG:core:db_free_rows: freeing 1 rows Sep 25 13:31:23 [24879] DBG:core:db_free_rows: row[0]=0x8187160 Sep 25 13:31:23 [24879] ERROR:core:do_action: bad uri <33179974674>, dropping packet Sep 25 13:31:23 [24879] DBG:core:parse_to: display={"9611375554"}, ruri={sip:9611375554@xyz.com} Sep 25 13:31:23 [24879] ERROR:core:parse_uri: bad uri, state 0 parsed: <3317> (4) / <33179974674> (25) Sep 25 13:31:23 [24879] ERROR:core:parse_sip_msg_uri: bad uri <33179974674(a)33.33.33.33> Sep 25 13:31:23 [24879] DBG:core:set_err_info: ec: 1, el: 3, ei: 'error parsing r-uri' Sep 25 13:31:23 [24879] ERROR:tm:new_t: uri invalid With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] Error compiling openser 1.2.2 notls on Ubuntu 8.04 x86_64
by Marcello Lupo 25 Sep '08

25 Sep '08

Hi, i'm getting this error on compiling openser-1.2.2-notls on Ubuntu 8.04 x86_64 Can you help me to see wath's is wrong? Same sources compiled perfect on SLES 10 x86_64. Thank you. Bye, Marcello gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops - falign-loops -ftree-vectorize -mtune=opteron -DNAME='"openser"' - DVERSION='"1.2.2-notls"' -DARCH='"x86_64"' -DOS='"linux"' - DCOMPILER='"gcc 4.2.3"' -D__CPU_x86_64 -D__OS_linux -D__SMP_yes - DCFG_DIR='"/data/openser/etc/openser/"' -DPKG_MALLOC -DSHM_MEM - DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE - DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT - DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN - DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL - DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT - DHAVE_SELECT -c action.c -o action.o In file included from parser/msg_parser.h:48, from action.h:27, from action.c:44: parser/../config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from parser/msg_parser.h:48, from action.h:27, from action.c:44: parser/../config.h:2:38: error: invalid digit "8" in octal constant parser/../config.h:2:41: error: invalid suffix "Z" on integer constant parser/../config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ parser/../config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ parser/../config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ parser/../config.h:19: error: expected identifier or ‘(’ before ‘if’ parser/../config.h:26:63: warning: missing terminating ' character parser/../config.h:26: error: missing terminating ' character In file included from action.c:45: config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from action.c:45: config.h:2:38: error: invalid digit "8" in octal constant config.h:2:41: error: invalid suffix "Z" on integer constant config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ config.h:19: error: expected identifier or ‘(’ before ‘if’ config.h:26:63: warning: missing terminating ' character config.h:26: error: missing terminating ' character In file included from route.h:32, from forward.h:39, from action.c:49: config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from route.h:32, from forward.h:39, from action.c:49: config.h:2:38: error: invalid digit "8" in octal constant config.h:2:41: error: invalid suffix "Z" on integer constant config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ config.h:19: error: expected identifier or ‘(’ before ‘if’ config.h:26:63: warning: missing terminating ' character config.h:26: error: missing terminating ' character In file included from ut.h:47, from action.c:54: config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from ut.h:47, from action.c:54: config.h:2:38: error: invalid digit "8" in octal constant config.h:2:41: error: invalid suffix "Z" on integer constant config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ config.h:19: error: expected identifier or ‘(’ before ‘if’ config.h:26:63: warning: missing terminating ' character config.h:26: error: missing terminating ' character In file included from mem/mem.h:35, from action.c:56: mem/../config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from mem/mem.h:35, from action.c:56: mem/../config.h:2:38: error: invalid digit "8" in octal constant mem/../config.h:2:41: error: invalid suffix "Z" on integer constant mem/../config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ mem/../config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ mem/../config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ mem/../config.h:19: error: expected identifier or ‘(’ before ‘if’ mem/../config.h:26:63: warning: missing terminating ' character mem/../config.h:26: error: missing terminating ' character In file included from mem/mem.h:54, from action.c:56: mem/f_malloc.h:134: warning: ‘struct mem_info’ declared inside parameter list mem/f_malloc.h:134: warning: its scope is only this definition or declaration, which is probably not what you want action.c: In function ‘do_action’: action.c:387: error: ‘ser_error’ undeclared (first use in this function) action.c:387: error: (Each undeclared identifier is reported only once action.c:387: error: for each function it appears in.) action.c:564: error: ‘rlist’ undeclared (first use in this function) make: *** [action.o] Error 1

2 3

[Kamailio-Users] Bind Access To Users to their IPs
by Ali Jawad 25 Sep '08

25 Sep '08

Hi I want some of my users only to be able to logon through a certain IP, is there some ACL mechanism that can be used to do that ? With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 4

[Kamailio-Users] SUBSCRIBE message with a private header
by Yoo Chan Jeon 24 Sep '08

24 Sep '08

Hi, I have a client sip agent which sends a SUBSCRIBE msg with a private header to the Kamailio presence server. Hwo do I get the private header informtaion? I want to read it and send it with a Notify message. Thanks. Jeon

2 1

[Kamailio-Users] sipgrep as non-privileged user
by Robert McNaught 24 Sep '08

24 Sep '08

Hi, I am using sipgrep to do sip traces on an openser/asterisk. I am find this great for getting readable traces - previously ngrep was difficult to read. However, to use sipgrep, I need to be logged in as the linux root user. I can change the PATH of an admin user so that they can have access to the command but they cannot use it as they need to be root in order to gain access to see the network traffic. Does anyone know of a way to do this? Is it normal to have engineers continually log in as root to take these traces - it seems awful dangerous? I know you could log in as a non-prileged user and put in sudo sipgrep -f xxx -t xxx, but not sure if people are actually doing this in production systems, or have a workaround? Thanks Robert

3 2

[Kamailio-Users] new feature: generic shared container in config language
by Elena-Ramona Modroiu 24 Sep '08

24 Sep '08

Hi, I just committed a new module: htable http://www.kamailio.org/docs/modules/devel/htable.html It adds a generic hash table container for usage in configuration file. The hash table is in shared memory, therefore the values are global over all kamailio processes. The items in hash table can be accessed via $sht(name). The name can include pseudo-variables that will be evaluated at runtime. In this way, the hash table can be used to store data per user and can offer functionality of dealing with array of values. An example of how to protect against dictionary attacks is in the README (link to online version above). As another example, loading values in hash tables from AVPs and building an array: $var(i) = 0; while($(avp(s:tmp)[$var(i)]) != NULL) { $sht(tmp[$var(i)]) = $(avp(s:tmp)[$var(i)]); $var(i) = $var(i)+1; } As a roadmap for the new module, planned for addition: - MI commands to get/set items in HT - ability to save/load in/from database - ability to init items at startup - auto-expire time for items This completes global variables space along with shared variables ($shv(name)) which are single value and allow only static name, but they are faster to access (see cfgutils readme file for more details on $shv(name)). Regards, Ramona -- Elena-Ramona Modroiu Web: http://www.asipto.com

3 2

[Kamailio-Users] Access List
by michel freiha 24 Sep '08

24 Sep '08

Hi All, Hi All,I need please to configure kamailio in order to force the user to register to my it from a specific IP...It's like creation of an access list...If the customer tries to register from another IP frm the one specified in kamailio he'll be denied Regards

3 2

[Kamailio-Users] digest or not digest ...
by BERGANZ François 24 Sep '08

24 Sep '08

Hello all, I need to choose if my users do digest auth or not! If I have : modparam("auth_db", "calculate_ha1", yes)àauth or modparam("auth_db", "calculate_ha1", 0)àauth digest! How can I choose if this user is digest after having look at a param in the database For example, user 1 have a column in the database which say ok he can do digest auth How can I ask to the client an auth digest ? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

2 2

[Kamailio-Users] openser.org and project documentation
by Jeremy McNamara 23 Sep '08

23 Sep '08

Whoever has taken over openser.org should do the community as a favor and at the very least HTTP 301 redirect the documentation links over to kamailio.org. There are soo many sites that index URLs like: http://openser.org/docs/modules/ ...however openser.org gives back a 404 not found.

1 0

[Kamailio-Users] About Changing URI's
by Ali Jawad 23 Sep '08

23 Sep '08

Hi I have the following setup Cell Phone "Any Possible Number" ----Calls a DID number-----> DID Provider --> Forwards Call to my Openser --> OpenSer Server--> Forwards Call to Asterisk with A2B --> Forwards it to the cell phone of the user In short, a person can call a DID and it gets forwarded to Openser, if the user is online using his sip phone, the sip phone rings and all is OK. However, if the user is not online I do forward that call to Asterisk and I want to forward it to the cell phone of the user. The problem is that the FROM field and the SOURCE URI keep changing,they are in the from phonenumberOfCaller(a)DID_Service_Provider.com and I do want to charge the user for the call. To do that I need to have the FROM field and the source URI to be the USER ID so that I can charge him for the call. So, I might have to change the source URI and FROM field of the packet and send it to asterisk..it would send the packet back to Openser and I have to change those back to what they were and then send it to the DID provider. Might sound confusing, and it might sound stupid. Please help me set this straight, can it be done ? Maybe not ? Maybe Yes... Any suggestions are welcome. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 1

[Kamailio-Users] un-subscribing a pending subscription is also pending
by jh＠tutpro.com 23 Sep '08

23 Sep '08

a user complained that un-subscribing a pending subscription is also pending, i.e., kamailio responds with 202 instead of 200. i verified the behavior. now the question is, what do the relevant RFCs say about it? it appears logical to me that if a user gives up waiting to get his/her subscription confirmed, he/she should be able to forget the whole thing by sending an un-subscribe. any comments on this? -- juha

2 2

[Kamailio-Users] CDRTool 6.6.3/6.6.4 with freeradius-cdrtool-1.1
by Diego Zuaneti Arruda 23 Sep '08

23 Sep '08

Hi all, it is possible to use CDRTool 6.6.3 or 6.6.4 with freeradius-cdrtool-1.1 ? The problem is in the code of freeradius-cdrtool.py, where in the code have "DebitBalance From=%s To=%s Duration=%s\n" , "MaxSessionTime From=%s To=%s Duration=%s Lock=1\n" and others, but the newer CDRTool need have a gateway parameter and Call Id parameter. Anybody have a sucess with this versions ? Thanks in advanced, Diego.

2 2

[Kamailio-Users] how to change source number
by BERGANZ François 23 Sep '08

23 Sep '08

Hello, I need to change the source number, to have an international format (e164) but when I use : $avp(s:245)=$fu; avp_subst("$avp(s:245)/$avp(s:246)/d", "/sip:/sip:33/"); uac_replace_from("", "$avp(s:246)"); in fact, my phone have in the screen sip:33..., before, it have my name I think that I have to use another function to change the source number, but which function? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

3 6

Re: [Kamailio-Users] nested redirect
by Daniel-Constantin Mierla 23 Sep '08

23 Sep '08

Hello, what you sent is the debug from kamailio/openser, which is useful, but I would need the sip messages, you can use ngrep for that, do ngrep -d any -qt -W byline port 5060 on your openser/kamailio server and send the output. Thanks, Daniel On 09/17/08 02:22, Kent Pirlo wrote: > ... is my core switch, > ... is my openser/Kamailio redirect server > ... is a redirect server from a carrier of ours that returns a billing TN in contact header.. > > > Here is the debug of the incoming ACK from the core switch after being sent the 300... > > > Sep 16 15:49:50 [6505] DBG:core:parse_msg: SIP Request: > Sep 16 15:49:50 [6505] DBG:core:parse_msg: method: <ACK> > Sep 16 15:49:50 [6505] DBG:core:parse_msg: uri: <sip:...@...> > Sep 16 15:49:50 [6505] DBG:core:parse_msg: version: <SIP/2.0> > Sep 16 15:49:50 [6505] DBG:core:parse_headers: flags=2 > Sep 16 15:49:50 [6505] DBG:core:parse_to_param: tag=391854ebdd4d0c63cf50836cbb12191e-6708 > Sep 16 15:49:50 [6505] DBG:core:parse_to: end of header reached, state=29 > Sep 16 15:49:50 [6505] DBG:core:parse_to: display={}, > Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: <To> [75]; > Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: to body > Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: cseq <CSeq>: <1> <ACK> > Sep 16 15:49:50 [6505] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKdce3ac34730f843a5a9b9c7002690f22>; state=16 > Sep 16 15:49:50 [6505] DBG:core:parse_via: end of header reached, state=5 > Sep 16 15:49:50 [6505] DBG:core:parse_headers: via found, flags=2 > Sep 16 15:49:50 [6505] DBG:core:parse_headers: this is the first via > Sep 16 15:49:50 [6505] DBG:core:receive_msg: After parse_msg... > Sep 16 15:49:50 [6505] DBG:core:receive_msg: preparing to run routing scripts... > Sep 16 15:49:50 [6505] DBG:sl:sl_filter_ACK: to late to be a local ACK! > Sep 16 15:49:50 [6505] DBG:core:destroy_avp_list: destroying list (nil) > Sep 16 15:49:50 [6505] DBG:core:receive_msg: cleaning up > > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, September 16, 2008 5:10 AM > To: kentp(a)globalpops.com > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] nested redirect > > Hello, > > can you post here the SIP messages? Will help to spot if there is > anything wrong with the ACK or something else in the signaling. > > Cheers, > Daniel > > > On 09/13/08 04:11, kentp(a)globalpops.com wrote: > >> i have a simple redirect server working: >> >> Call comes from switch (A) to kamailio redirect server (B).. >> (B) runs perl script to determine route >> (B) sends Stateless 300 redirect reply to (A). >> (A) sends ACK to (B). >> >> Now, I need to add a nested redirect... >> >> call comes from sitch (A) to kamailio redirect server (B).. >> (B) then needs to send the call to third party proxy (C).. >> (C) will return 302 redirect.. >> (B) sends ACK to (C) >> (B) needs to read the contact header, then run perl script to determine route (using the contact value >> returned from (c)) >> (B) then sends 300 redirect back to (A). >> (A) sends ACK back to (B) >> >> >> so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending >> multiple 300's to (A): >> >> route { >> route(1); >> } >> >> route[1] { >> t_on_reply("1"); >> t_on_failure("2"); >> if (!t_relay("proxy.server.c:5060","0x02")) { >> sl_reply_error(); >> } >> onreply_route[1] { >> if(t_check_status("302")){ >> $var(lrn_from_bv) = $ct; >> } >> } >> failure_route[2] { >> >> if (t_check_status("3[0-9][0-9]")) { >> route(2); >> exit; >> } >> route[2] { >> if(perl_exec("callcheck")){ >> avp_pushto("$ru", "$avp(s:routes)/g"); >> t_reply("300","Go Here"); >> exit; >> } >> } >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] Question about redirect if user is not online
by Ali Jawad 23 Sep '08

23 Sep '08

Hi All I do forward calls to a DID number to my openser box, if my softphone is online it rings directly because I am using the DID number as an alias. However I want the call to be forwarded to my asterisk box if the user is not online. What statement should I use ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 7

[Kamailio-Users] preparing release 1.4.1
by Daniel-Constantin Mierla 22 Sep '08

22 Sep '08

Hello, I propose to release kamailio 1.4.1 next Thursday, Sep 25, 2008. Please send to devel(a)lists.kamailio.org any critical issues you are aware of, to try to fix them. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] Handling server with broken record-route handling
by Marcus Better 22 Sep '08

22 Sep '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have problems with a remote server, outside of my control, that seems to have very broken request URI handling. It is possibly suffering from an Asterisk bug [1] or something similar. On my side I have a UAC (IP address 172.16.1.102) and a SER 2.0 proxy. The UAC sends an INVITE, and the remote UAS ends the dialog with a BYE request. However this BYE returns to my proxy with missing route headers, so SER cannot route it. I need to work around the problem with SER on my side and wonder what the best solution is. Here is the relevant part of the conversation as seen on my proxy 80.81.82.3 (slightly massaged, irrelevant parts cut): INVITE sip:00463112341234@proxy.remote.com SIP/2.0. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 INVITE. To: <sip:00463112341234@proxy.remote.com>. From: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. Via: SIP/2.0/UDP 80.81.82.3;branch=z9hG4bK761b.b1548ac4.0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK74592b710287e6cfa94b4cf9c10e15bf. Contact: "17473605948" <sip:172.16.1.102:5060;transport=udp>. SIP/2.0 200 OK. Via: SIP/2.0/UDP 80.81.82.3;branch=z9hG4bK761b.b1548ac4.0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK74592b710287e6cfa94b4cf9c10e15bf. Record-Route: <sip:123.12.13.14;lr;ftag=2045565953650740630>. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. From: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. To: <sip:00463112341234@proxy.remote.com>;tag=as438ee174. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 INVITE. Contact: <sip:011463112341234@130.94.88.93:5060>. ACK sip:01146739835569@130.94.88.90:5060 SIP/2.0. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 ACK. Via: SIP/2.0/UDP 80.81.82.3;branch=0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK96a96b64dfd5aba5c896c37e0de18850. From: <sip:17473605944@proxy.remote.com>;tag=7737826714334905221. To: <sip:0046739835569@proxy.remote.com>;tag=as543c9f3f. Route: <sip:123.12.13.14;lr;ftag=7737826714334905221. Content-Type: application/sdp. BYE sip:80.81.82.3:5060 SIP/2.0. Record-Route: <sip:123.12.13.14;lr;ftag=as438ee174>. Via: SIP/2.0/UDP 123.12.13.14;branch=z9hG4bK100d.d617f387.0. Via: SIP/2.0/UDP 130.94.88.93:5060;branch=z9hG4bK0256ad7c;rport=5060. Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. From: <sip:00463112341234@proxy.remote.com>;tag=as438ee174. To: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 102 BYE. I think this BYE is incorrect. It should have the originator's Contact (sip:172.16.1.102:5060;transport=udp) in the request URI and the Record-Routes of the original INVITE in the Route header. Or, if it is a strict router, it could have the current request URI but add the Contact as the last Route header. Now it has destroyed the originator's route information. What is the best way to get SER to remember/reconstruct this last hop to the UAC? Cheers, Marcus [1] http://bugs.digium.com/view.php?id=3609 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkjXkbQACgkQXjXn6TzcAQnhrQCeP9T0pR+AlPJjTr8OCS5nNOAA GBIAnR5pw68rqe1PR00jDLvcYh+eQXPP =sdca -----END PGP SIGNATURE-----

1 0

[Kamailio-Users] Kamailio+Asterisk Need Help
by VIP Carrier 22 Sep '08

22 Sep '08

Hi, I'm trying to setup an HA cluster based of 2 Proxy 4 Asterisk 2 DB's Can some one please help me with this.

6 8

[Kamailio-Users] Crate prepaid account on CDRTool
by Adrian Georgescu 20 Sep '08

20 Sep '08

Click on the Rating link then select Prepaid table. There you can add new accounts Adrian

2 1

[Kamailio-Users] New MediaProxy release 2.1.0
by Adrian Georgescu 20 Sep '08

20 Sep '08

MediaProxy2 uses the Linux kernel for routing packets.So it uses all the cores and CPUs recognized by the kernel. Adrian

3 8

[Kamailio-Users] OT: openser/Asterisk management GUI
by Klaus Darilion 19 Sep '08

19 Sep '08

Hi! FYI: I just found this one: http://carrierclass.net/ looks like it is a central management webinterface for a openser-asterisk setup. regards klaus

3 3

[Kamailio-Users] Problem Dynamically Setting Forward
by Jon Farmer 18 Sep '08

18 Sep '08

Hi I am trying to call a script via exec_avp and then use the output of the script to use as an argument to forward(). Such as route[2] { # forward to AGCF exec_avp("/usr/utils/select_incoming_agcf.php"); record_route(); xlog("forward to: $avp(i:1)\n"); #forward("$avp(s:1)"); forward("$avp(s:1); exit; } However syslog complains as follows Sep 18 11:42:12 vla openser[19470]: ERROR:core:parse_phostport: bad protocol in $avp(s:1) Sep 18 11:42:12 vla openser[19470]: ERROR:core:fix_actions: ERROR:fix_actions: FORWARD/SEND bad argument Sep 18 11:42:12 vla openser[19470]: ERROR:core:main: failed to fix configuration with err code -6 The output of the script is 10.0.0.1:5060 Any ideas? Regards Jon

2 1

[Kamailio-Users] Using asterisk to pstn integration
by Tiago Rés 18 Sep '08

18 Sep '08

Hello all, I'm running openser 1.2.2, and i want to make/receive pstn calls. I have installed asterisk 1.4. in the same machine. Do I need to change the port in asterisk? and how to do it? What are the basic lines in the configuration files of asterisk to accomplish that? I don't think I have problem with openser.cfg. Sorry about the questions but I don't seem to find the exact documentation I need. Thanks in advance! Tiago Res

2 1

[Kamailio-Users] Crate prepaid account on CDRTool
by Diego Zuaneti Arruda 18 Sep '08

18 Sep '08

Hello, Somebody have a example to create a prepaid account on CDRTool database ? Thanks. Diego.

1 0

[Kamailio-Users] double resquests
by BERGANZ François 18 Sep '08

18 Sep '08

Hello, I need help because, when myopenser look at a frame, it forward it in double always. You can look at it on the join picture. Thank you for advices. Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

3 5

[Kamailio-Users] Help Me on OpenXCAP...
by mahesh.peddi＠info-spectrum.com 18 Sep '08

18 Sep '08

Hi ALL, I installed openxcap 1.0 version and its dependences. When i run "openxcap --no-fork" and openser 1.3.2v. I am getting these errors. ************************************************************************************************************* [root@info infopresence-1.2.0-tls]# openxcap --no-fork Starting Open XCAP 1.0.0 Traceback (most recent call last): File "/usr/local/bin/openxcap", line 53, in <module> from xcap.server import XCAPServer File "/usr/local/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> from xcap import authentication File "/usr/local/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> from xcap.appusage import getApplicationForURI, namespaces File "/usr/local/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 42, in <module> class ServerConfig(ConfigSection): File "/usr/local/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 45, in ServerConfig backend = Backend('Database') File "/usr/local/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 40, in __new__ raise ValueError("Couldn't find the '%s' backend module: %s" % (value.lower(), str(e))) ValueError: Couldn't find the 'database' backend module: No module named _mysql_exceptions [root@info infopresence-1.2.0-tls]# ******************************************************************************************************************** Here with i am enclosing the config.ini file. Please guide me on this. Thanks & Regards, Mahesh Peddi

1 0

[Kamailio-Users] Rtpproxy limit
by alasupcom 18 Sep '08

18 Sep '08

hi all :working: When i use force_rtp_proxy() in my openser.cfg i have 20 calls/s, if i try to increase the number i have this error: force_rtp_proxy2:incorrect port 0 in reply from rtp proxy and i can't use rtpproxy in this calls. But in this linkc http://www.b2bua.org/wiki/RTPproxy/FAQ i have the capacity rtpproxy 1500-2000. Please, if one have the solution send me a reply.:confused: any suggest is appreciate -- View this message in context: http://www.nabble.com/Rtpproxy-limit-tp19532035p19532035.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Kamailio-Users] RTPproxy
by Ricky Gutierrez 18 Sep '08

18 Sep '08

hi list rtpproxy version 1.1 runs well with kamailio 1.4 or openser 1.3.2 ?, some person in the list that this making tests with her Sep 17 21:55:26 xserver /sbin/kamailio[3522]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:26 xserver /sbin/kamailio[3531]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:26 xserver /sbin/kamailio[3535]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:26 xserver /sbin/kamailio[3518]: WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Sep 17 21:55:26 xserver /sbin/kamailio[3518]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:25 xserver /sbin/kamailio[3537]: ERROR:nathelper:send_rtpp_command: proxy <udp:127.0.0.1:7890> does not respond, disable it Sep 17 21:55:26 xserver /sbin/kamailio[3537]: WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Sep 17 21:55:26 xserver /sbin/kamailio[3537]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily I have some problems with the one regards rickygm

2 1

[Kamailio-Users] New MediaProxy release 2.1.0
by Adrian Georgescu 18 Sep '08

18 Sep '08

Hello, There is a new release of MediaProxy software available. For people running Debian testing or unstable on an i386 architecture there is now an official public repository provided by AG Projects. To use it, add these lines in /etc/apt/sources.list # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main Install the AG Projects debian software signing key: wget http://download.ag-projects.com/agp-debian-gpg.key apt-key add agp-debian-gpg.key After that, run: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions to install all the packages, or you can install only the packages you actually need on that specific system. The change log follows: mediaproxy (2.1.0) unstable; urgency=low . * Mention libnetfilter-conntrack in the installation requirements section * Added note about using distribution provided kernel images * Renamed OpenSER to OpenSIPs * Do accounting only after a 200 OK response was received, which is indicated by the presence of the to_tag that is not sent by OpenSIPS for provisional replies, but only with the 200 OK * Have synthetic test scripts look for the mediaproxy module in current and parent directories automatically * In the synthetic tests have OpenSIPS send the to tag only with 200 OK * Added synthetic test script to test provisional responses from two different PSTN gateways with varying to tags * Enhanced kernel version parsing to support non-standard named kernels (i.e. major.minor.revision.extra) * Updated debian standards version * Rephrased description in manual pages and debian control file * Restructured and improved documentation to better describe how to install * Improved dictionary parsing in RADIUS accounting module * Update support section in readme * Added management interface for OpenSIPS * Fix for relay cleanup timer when dispatcher is restarted * Modified dispatcher to store dialog_id for sessions and use the OpenSIPS management interface to send BYEs when a session did timeout * Added dialog_id value to test scripts * Simplified parsing of headers received from OpenSIPs in dispatcher * Fixed ExpireWatcher port comparison to be done as a 16-bit unsigned int * Remove all matching iptables entries on Inhibitor dealloc in order to remove stray entries that are leftover after crashes * Added per-stream post-dial delay to the session statistics * Record the post dial delay from the first stream into radius accounting * Added note to INSTALL about official Debian repositories from AG Projects * Added note to INSTALL about where to find the changelog * Updated the feature list in README with the latest additions * Fixed regression from 1.x that prevented chaining of mediaproxies * Ignore invalid accounting module names in the configuration and log an error about them instead of raising an exception * Gracefully report the errors that occur while reading the radius configuration file and then exit, instead of throwing an exception * The radius configuration file may be relative to the config directory * Do not cache database inserted rows to avoid the need to clean the cache or otherwise have an ever increasing application memory footprint * Capture and report database errors when inserting statistics records * Use MyISAM for the media_sessions table * Improved the database accounting speed about 3 times using raw queries * Create the table for database accounting on startup if it doesn't exist * Removed pool_size database option and use a single database connection * Reworded the tls README a bit for clarity * Added OpenSIPS section to the config.ini.sample * Added documentation sections for accounting and closing expired calls * Added minimum version dependency for libnetfilter-conntrack in INSTALL * Encode applications and codecs from all streams, not only the first one, in the radius accounting backend * Log the post-dial delay in radius accounting as Acct-Delay-Time * Capture and log radius exceptions For more information, download of source code and the documentation, visit the home page of the product: http://ag-projects.com/MediaProxy.html Regards, Adrian

3 2

[Kamailio-Users] Presence server (acive and stand-bye setup)
by Yoo Chan Jeon 18 Sep '08

18 Sep '08

Hi, I just start to use the Openser as the presence server. I tested with the eyebeam. The server saved the presentity information to the MySQL server. It worked well. I have questions. I would like to setup two presence servers. One is active, and the other is stand-bye. If the active is down, the client sends a message to the stand-bye. How does the presence server know that it is active or stand-bye? How do I set it up? How do I replicate the database between servers? If the active is down, how do I make it up running again? Thanks. Jeon

2 1

[Kamailio-Users] CDRTool 6.6.3 not rating calls
by Adrian Georgescu 18 Sep '08

18 Sep '08

The data in your tables seems to be consistent. Why you still do not get the price is not clear from your email. Adrian

1 0

[Kamailio-Users] Issues with routing
by Jon Farmer 17 Sep '08

17 Sep '08

Hi all I have to following setup for a incoming call Asterisk -> OpenSER -> 3rd Party SBC So the call comes into Asterisk which sends it to OpenSER which forwards it to the 3rd party SBC. I need to use a forward() to send to the SBC as the URI needs to be preserved as it came from Asterisk. This is all working fine, however when the called party answers and the OK is sent for the INVITE it is not sent back to Asterisk. How do I get this to work? Regards Jon

2 7

[Kamailio-Users] CDRTool 6.6.3 not rating calls
by Diego Zuaneti Arruda 17 Sep '08

17 Sep '08

Hi, The CDRTool is not generate rating and prices. My configuration: -- billing_rates | id | gateway | domain | subscriber | name | destination | durationRate | trafficRate | application | connectCost | +----+---------+--------+------------+------+-------------+--------------+-------------+-------------+-------------+ | 1 | | | | 551 | 55 | 1500 | | audio | 0 | | 2 | | | | 552 | 55 | 1000 | | audio | 0 | -- billing_profiles | id | gateway | domain | subscriber | name | rate_name1 | hour1 | rate_name2 | hour2 | rate_name3 | hour3 | rate_name4 | hour4 | +----+---------+--------+------------+------+------------+-------+------------+-------+------------+-------+------------+-------+ | 1 | | | | 551 | 551 | 18 | 552 | 0 | | 0 | | 0 | | 2 | | | | 552 | 552 | 18 | 552 | 0 | | 0 | | 0 | -- billing_customers | id | gateway | domain | subscriber | profile_name1 | profile_name1_alt | profile_name2 | profile_name2_alt | timezone | increment | min_duration | country_code | +----+---------+-------------------+------------+---------------+-------------------+---------------+-------------------+-------------------+-----------+--------------+--------------+ | 1 | | mydomain.com | | 551 | | 552 | | America/Sao_Paulo | 0 | 0 | | -- destinations | id | gateway | domain | subscriber | dest_id | dest_name | +----+---------+--------+------------+---------+-----------+ | 1 | | | | 55 | Brazil | -- And the radacct: | 51 | 1vpzOG09ZNwBth5S(a)192.168.2.29 | 695fa85e025f57eb | 87002(a)mydomain.com | mydomain.com | | 5060 | | 2008-09-17 18:05:58 | 2008-09-17 18:06:03 | 5 | NULL | NULL | | 0 | 0 | 00551533334444(a)mydomain.com | 87002(a)mydomain.com | 200 | Sip-Session | | | 0 | 0 | NULL | Invite | 200 | as7e8c15b3 | gluy7ejee3XZ1ckO | =2B551533334444(a)200.200.200.200 | | audio | | | | 201.201.201.201 | 5060 | 00551533334444(a)mydomain.com | | 0 | | | 0.0000 | 1 | | NULL | | | | | The configuration is correct? Thanks. Diego.

1 0

[Kamailio-Users] digest authentication
by BERGANZ François 17 Sep '08

17 Sep '08

Hello, I need help to do digest authentication ! I have: Route{ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "db_url", "mysql://root:xxx@localhost/openser") if (!proxy_authorize("realm.org", "subscriber")) { proxy_challenge("realm.org", "1"); exit; }; } And, in my database : username domain password realm Ha1 gaia Realm.org xxxxx Where can be my problem ? Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

1 1

[Kamailio-Users] new modules for dialog-info
by Klaus Darilion 17 Sep '08

17 Sep '08

Hi! I have just commited 2 new modules into SVN trunk: - pua_dialoginfo: gets dialog states from dialog module and sends PUBLISH with the dialog state in RFC 4235 conform XML documents - presence_dialoginfo: extends presence module to handle PUBLISH with dialog-info. Allows aggregation of multiple XML documents into one document. With this module you can have BLF without the need for an PBX (Asterisk). It is know to work with SNOM and Linksys phones. Linksys requires some tweaking with module parameters. The work was sponsored by Silver Server (www.sil.at) For a full description please refer to the very extensive documentation in the associated READMEs. regards Klaus

4 3

Re: [Kamailio-Users] t_relay
by Daniel-Constantin Mierla 17 Sep '08

17 Sep '08

sabino frisardi wrote: > OK, so if i call tmb->t_relay(msg,0,0,0,0,0,0) from a module it's the > same if i call t_relay() from script? yes. Daniel > > 2008/9/3 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 09/02/08 13:20, Klaus Darilion wrote: > > Hi Sabino! > > Do you really need to call it from within your module? The > usual way is that modules prepare the request (set RURI, DURI, > branches ...) and then in script you call t_relay(). This way > is more flexible as it also allows you to perform some task in > script before sending out the request (if needed). > > klaus > > sabino frisardi schrieb: > > > Hi all, > > I'm using openser 1.4. Can anyone tell me how can i call > t_relay() from another module. > Can anyone help me? any suggestion? > > > if you need it, then tm module exports an API that you can use. > You can have a look at msilo module to see how the TM api is > imported and used later. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > > > -- > Sabino Frisardi -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Iam getting ERROR:db_mysql:db_mysql_new_connection: driver error
by mahesh.peddi＠info-spectrum.com 16 Sep '08

16 Sep '08

Hi, I am working on opensips 1.4.0 and while running "openser -C" command I am getting the following errors. Please any one can help me. Sep 16 21:07:05 localhost opensips[3200]: ERROR:db_mysql:db_mysql_new_connection: driver error: Access denied for user 'opensips'@'localhost' (using password: YES) Sep 16 21:07:05 localhost opensips[3200]: ERROR:core:db_do_init: could not add connection to the pool Sep 16 21:07:05 localhost opensips[3200]: ERROR:rls:mod_init: while connecting database Sep 16 21:07:05 localhost opensips[3200]: ERROR:core:init_mod: failed to initialize module rls Sep 16 21:07:05 localhost opensips[3200]: ERROR:core:main: error while initializing modules Sep 16 21:07:05 localhost opensips[3200]: NOTICE:presence:destroy: destroy module ... Sep 16 21:57:10 localhost yum-updatesd: error getting update info: Cannot find a valid baseurl for repo: extras Waiting for your replies. Thanks & Regards, Mahesh

2 1

[Kamailio-Users] nested redirect
by kentp＠globalpops.com 16 Sep '08

16 Sep '08

i have a simple redirect server working: Call comes from switch (A) to kamailio redirect server (B).. (B) runs perl script to determine route (B) sends Stateless 300 redirect reply to (A). (A) sends ACK to (B). Now, I need to add a nested redirect... call comes from sitch (A) to kamailio redirect server (B).. (B) then needs to send the call to third party proxy (C).. (C) will return 302 redirect.. (B) sends ACK to (C) (B) needs to read the contact header, then run perl script to determine route (using the contact value returned from (c)) (B) then sends 300 redirect back to (A). (A) sends ACK back to (B) so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending multiple 300's to (A): route { route(1); } route[1] { t_on_reply("1"); t_on_failure("2"); if (!t_relay("proxy.server.c:5060","0x02")) { sl_reply_error(); } onreply_route[1] { if(t_check_status("302")){ $var(lrn_from_bv) = $ct; } } failure_route[2] { if (t_check_status("3[0-9][0-9]")) { route(2); exit; } route[2] { if(perl_exec("callcheck")){ avp_pushto("$ru", "$avp(s:routes)/g"); t_reply("300","Go Here"); exit; } }

2 1

[Kamailio-Users] OpenSER 1.3 crash
by Robin Vleij 16 Sep '08

16 Sep '08

Hi guys! Posted this to the dev list, but I'm not sure if that's the right forum. Looking at the posts there, it's more about patches and dev stuff (hence the name). Hoped to get some feedback on a crash I had a week ago, now I'm trying it here. :) We've been running OpenSER (1.3.0 ntls, compiled for x66_64) for quite a while now and only had a crash once. We suspected then that this was caused by hitting the max_while_loops for a while (never happened again after fixing the max-while-loops). Now I had a crash and the backtrace looks like the following. Do you guys have any pointers on what is causing this? #0 free_hostent (dst=0x6e3fe0) at proxy.c:203 203 for (r=0; dst->h_addr_list[r];r++) { (gdb) backtrace #0 free_hostent (dst=0x6e3fe0) at proxy.c:203 #1 0x0000000000436530 in free_proxy (p=0x6e3fc0) at proxy.c:312 #2 0x00002adec6dc01ea in t_relay_to (p_msg=0x6de4a0, proxy=0x6e3fc0, flags=<value optimized out>) at t_funcs.c:230 #3 0x00002adec6dd1fcd in w_t_relay (p_msg=0x6de4a0, proxy=0x0, flags=0x0) at tm.c:929 #4 0x000000000040ba9f in do_action (a=0x646e80, msg=0x6de4a0) at action.c:816 #5 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #6 0x000000000040ca06 in do_action (a=0x646fb0, msg=0x6de4a0) at action.c:689 #7 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #8 0x000000000040d780 in do_action (a=0x647460, msg=0x6de4a0) at action.c:695 #9 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #10 0x000000000040ca06 in do_action (a=0x6475a0, msg=0x6de4a0) at action.c:689 #11 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #12 0x000000000040e309 in run_top_route (a=0x643398, msg=0x6de4a0) at action.c:112 #13 0x000000000044b8bf in receive_msg ( buf=0x625ca0 "ACK sip:013104104@sip-corporate.tele2.se SIP/2.0\r\nVia: SIP/2.0/UDP 130.244.190.46;branch=z9hG4bK7b7c7b36\r\nVia: SIP/2.0/UDP 130.244.190.46;branch=z9hG4bK7b7c7b36\r\nVia: SIP/2.0/UDP 130.244.190.46;branch"..., len=3703, rcv_info=0x7fffe4d8d840) at receive.c:156 #14 0x0000000000488154 in udp_rcv_loop () at udp_server.c:438 #15 0x0000000000425081 in main (argc=9, argv=0x7fffe4d8da48) at main.c:834

2 1

[Kamailio-Users] help to query a table with new function in core
by Marcio 15 Sep '08

15 Sep '08

Hi all, I need a help to solve some problems in the clearance of a function to perform a query Mysql....I created a function based on documentation of asipto / kamailio devel which is very good... But as I did not want to create a new module I'm trying to use the module domain... whose goal a proxy SIP receive msg that will should validate the realm comparing with a table set up in MySQL(trusdomain)... it should be used into config file to to declare of the table created in the Mysql: modparam ( "domain", "trusdomain_table," trusdomain) the version table of the openser has updated with the creation of this new table (trusdomain) and created a function into file: domain.c / h and a part was need be exported to config file in domain_mod.c / h....but on the tests a call..this proxy kills the pid openser ... Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: child process 7964 exited by a signal 11 Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: core was generated Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: terminating due to SIGCHLD Sep 5 06:32:21 notehome openser[7967]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7963]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7968]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7966]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7965]: INFO:core:sig_usr: signal 15 received The function created in domain.c: int is_from_trusted(struct sip_msg* _msg, char* _s1, char* _s2) { struct sip_uri *puri; /* struct str* _host; */ /* str table_trus; str col_id; str col_dom; str col_lastmod; */ str* _host = (str*)(&(puri->host)); char* db_url = DEFAULT_RODB_URL; db_key_t keys[1]; /* Tipo da coluna .. Nome das colunas */ db_val_t vals[1]; db_key_t cols[1]; db_res_t* res = NULL; if ((puri=parse_from_uri(_msg))==NULL){ LM_ERR("Error while parsing From header\n"); return -2; } /* Bind the database module */ if (bind_dbmod(db_url, &domain_dbf )) { LM_ERR("Cannot bind to database module!\n"); return -1; } /* Check for SELECT capability */ if (!DB_CAPABILITY(domain_dbf, DB_CAP_QUERY)){ LM_ERR("Database modules does not provide all functions needed\n"); return -1; } /* Connect to DB */ db_handle = domain_dbf.init(db_url); if (!db_handle){ LM_ERR("failed to connect database\n"); return -1; } keys[0]=trusdomain_col.s; cols[0]=trusdomain_col.s; LM_INFO(">>domain.c>>-2-IF RES_ROW_N(res)=>%d\n",RES_ROW_N(res)); /* Estrutura dos vls das colunas - arq. db/db_val.h */ VAL_TYPE(vals) = DB_STR; VAL_NULL(vals) = 0; VAL_STR(vals).s = _host->s; VAL_STR(vals).len = _host->len; /* LM_INFO(">>domain.c>>-1-connector=%d and tab=%s\n",db_handle,trusdomain_table.s); */ if (domain_dbf.use_table(db_handle, trusdomain_table.s) < 0) { LM_ERR("Error while trying to use trusdomain table\n"); return -1; } if (domain_dbf.query(db_handle, keys, NULL, vals, cols, 1, 1, 0, &res) < 0) { LM_ERR("Error while querying database\n"); domain_dbf.close(db_handle); return -1; } /* RES_ROW_N(res) Num. de linhas no fetch */ if (RES_ROW_N(res) <= 0 || RES_ROWS(res)[0].values[0].nul != 0){ LM_DBG("no value found\n"); if (res != NULL && domain_dbf.free_result(db_handle, res) < 0) LM_DBG("failed to free the result\n"); domain_dbf.close(db_handle); } else if (RES_ROW_N(res) >= 1) { /* Teste do que estah na tab. trusdomain com o que vem na URI */ LM_DBG("Realm '%.*s' is trusted\n",_host->len, ZSW(_host->s)); } /* Free the result */ domain_dbf.free_result(db_handle, res); domain_dbf.close(db_handle); return 0; } And declaration in domain_mod.c : ... /* by mrg */ #define TRUSDOMAIN_TABLE "trusdomain" #define TRUSDOMAIN_TABLE_LEN (sizeof(TRUSDOMAIN_TABLE) - 1) #define TRUSDOMAIN_COL "trusdomain" #define TRUSDOMAIN_COL_LEN (sizeof(TRUSDOMAIN_COL) - 1) ... ... str trusdomain_table = {TRUSDOMAIN_TABLE, TRUSDOMAIN_TABLE_LEN}; /* Name of trusted table */ str trusdomain_col = {TRUSDOMAIN_COL, TRUSDOMAIN_COL_LEN}; /* Name of trusted column */ ... ... /* * Exported functions */ static cmd_export_t cmds[] = { ............. {"is_domain_local", w_is_domain_local, 1, parameter_fixup, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {"is_from_trusted", is_from_trusted, 0, 0, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {0, 0, 0, 0, 0, 0} }; .../* * Exported parameters */ static param_export_t params[] = { ............ {"domain_table", STR_PARAM, &domain_table.s}, {"domain_col", STR_PARAM, &domain_col.s }, {"trusdomain_table", STR_PARAM, &trusdomain_table.s}, {"trusdomain_col", STR_PARAM, &trusdomain_col.s}, {0, 0, 0} }; ... ... static int mod_init(void) { int vertr....; ..... trusdomain_table.len = strlen(trusdomain_table.s); trusdomain_col.len = strlen(trusdomain_col.s); ..... if (db_mode != 0) { if (domain_db_init(db_url.s)<0) return -1; /* Check table version */ ver = domain_db_ver(&domain_table); vertr = domain_db_ver(&trusdomain_table); if ((ver < 0) && (vertr < 0)) { LM_ERR("Error while querying table version\n"); goto error; } else if ((ver < TABLE_VERSION) && (vertr < TABLE_VERSION)) { LM_ERR("Invalid table version <%d> trustable version <%d> (should be %d)\n", ver, vertr, TABLE_VERSION); goto error; } ....... ....... /* Initializing hash tables and hash table variable */ ........ domain_db_close(); } return 0; error: domain_db_close(); return -1; } So I think that I must be something doing wrong ..... as would be the right and as I could learn fast to resolve error debugging of pID of Openser that has been deads...??? thx Marcio

2 2

[Kamailio-Users] project updates
by Daniel-Constantin Mierla 15 Sep '08

15 Sep '08

Hello everybody, just want to give some updates concerning two main topics. First, regarding the project name, many expressed the concern that a rename in short time will create more confusion. As we are not in front of a new major release, I think it is better to postpone this process for a while and focus on the second topic, the organization for the project. For this one, I have to thank personally to many people from the list that wrote us, giving suggestions and helping finding solutions. As a result, we are working to create the draft of statute for organization, but we are also investigating the option to join an existing organization, dedicated to hosting of open source projects. We will keep you updated. Meanwhile, the svn repository got couple of new features as well as refactoring of some modules. 1.4.1 will be released soon as we got a bunch of important fixes in the 1.4 branch. Most probably another release in 1.3 branch will follow. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] record_route to control my cancel.
by BERGANZ François 15 Sep '08

15 Sep '08

Hello, I dont really understand, but : I just need to do in the INVITE : record_route_preset('$dd:$dp;codec=alaw'); forward(); and for the CANCEL: if(check_route_param(";codec=alaw")){ route(2); } Is it possible? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net

5 6

[Kamailio-Users] Question about Authentication
by mark morreny 15 Sep '08

15 Sep '08

Hi, I am able to get authentication setup with Mysql and Freeradius. Just one simple question, for pure authentication purpose, is it better just to use the Mysql database for authentication instead of setting up Freeradius? Is there any advantage of using Freeradius on top of Mysql? If I use Mysql to serve authentication for Openser, is it possible to encrypt the password in Mysql? Is it possible to specify the table and fields to be used for authentication instead of using the default mysql tables? Thanks for your help. Regards, Mark

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[Kamailio-Users] DNS_name to IP coredumps on calee bye
by Stefan Broddesson 15 Sep '08

15 Sep '08

Greetings, Rev. is 1.3.3 on debian_Etch_64 Have a scenarion that always triggers a core dump. On Calee initiated <bye> and where destination address is a hostname not IP address. Executing forward() or t_relay() in script triggers the coredump. dns & rev_dns On/Off with the same result. --- DBG:core:comp_scriptvar: str 20 : siphost.domain.com DBG:tm:t_newtran: transaction on entrance=0xffffffffffffffff DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:parse_headers: flags=78 DBG:tm:t_lookup_request: start searching: hash=7994, isACK=0 DBG:tm:matching_3261: RFC3261 transaction matching failed DBG:tm:t_lookup_request: no transaction found DBG:tm:run_reqin_callbacks: trans=0x2b4e6b2e0f48, callback type 1, id 0 entered DBG:siptrace:trace_onreq_in: trace off... DBG:core:mk_proxy: doing DNS lookup... DBG:core:sip_resolvehost: no port, no proto -> do NAPTR lookup! DBG:core:filter_and_sort_naptr: found valid SIP+D2U -> _sip._udp.siphost.domain.com DBG:core:filter_and_sort_naptr: skipping SIPS+D2T -> _sips._tcp.siphost.domain.com DBG:core:filter_and_sort_naptr: skipping SIP+D2T -> _sip._tcp.siphost.domain.com Have a workaround that rewrites destination in case of hostName trig. impractical though. -hints? //s __________________________________________________________ Ta semester! - sök efter resor hos Kelkoo. Jämför pris på flygbiljetter och hotellrum här: http://www.kelkoo.se/c-169901-resor-biljetter.html?partnerId=96914052

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[Kamailio-Users] MediaProxy 2.0.3 on OpenSUSE 11
by Darren Sessions 15 Sep '08

15 Sep '08

Just finished some documentation on how to install MediaProxy 2.0.3 on OpenSUSE 11. Due to the length of the post and the init scripts provided, I won't be re-posting this one in the lists like usual. Thanks do go out to Ruud Klaver @ AG Projects for a clever Linux version detection hack for the relay module (hopefully will make it into the next release of MediaProxy). Everything can be found on my site at http://www.darrensessions.com. Cheers! _____________________________ Darren Sessions dmsessions(a)gmail.com http://www.darrensessions.com _____________________________

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[Kamailio-Users] Help to create in core
by Marcio 13 Sep '08

13 Sep '08

Hi all, I appreciate very much to help solve some problems in the clearance of a function to perform a query Mysql .... created the function based on documentation of asipto / kamailio devel which is great ... but I did not want to create a new module used the module in domain ... whose goal is a proxy to receive SIP msg should validate the realm with what is in a table (Trustee) was created through a query ... it should be used for the config file to declare of the table created in the Mysql: modparam ( "domain", "trusdomain_table," trusdomain) the table version updated with the creation of this table (trusdomain) and create the own function in the file: domain.c / h part to be exported in domain_mod.c / h....but on the tests a call SIP to proxy kills a openser's pid ... Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: child process 7964 exited by a signal 11 Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: core was generated Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: terminating due to SIGCHLD Sep 5 06:32:21 notehome openser[7967]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7963]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7968]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7966]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7965]: INFO:core:sig_usr: signal 15 received *The function created in domain.c:* int is_from_trusted(struct sip_msg* _msg, char* _s1, char* _s2) { struct sip_uri *puri; /* struct str* _host; */ /* str table_trus; str col_id; str col_dom; str col_lastmod; */ str* _host = (str*)(&(puri->host)); char* db_url = DEFAULT_RODB_URL; db_key_t keys[1]; /* Tipo da coluna .. Nome das colunas */ db_val_t vals[1]; db_key_t cols[1]; db_res_t* res = NULL; if ((puri=parse_from_uri(_msg))==NULL){ LM_ERR("Error while parsing From header\n"); return -2; } /* Bind the database module */ if (bind_dbmod(db_url, &domain_dbf )) { LM_ERR("Cannot bind to database module!\n"); return -1; } /* Check for SELECT capability */ if (!DB_CAPABILITY(domain_dbf, DB_CAP_QUERY)){ LM_ERR("Database modules does not provide all functions needed\n"); return -1; } /* Connect to DB */ db_handle = domain_dbf.init(db_url); if (!db_handle){ LM_ERR("failed to connect database\n"); return -1; } keys[0]=trusdomain_col.s; cols[0]=trusdomain_col.s; LM_INFO(">>domain.c>>-2-IF RES_ROW_N(res)=>%d\n",RES_ROW_N(res)); /* Estrutura dos vls das colunas - arq. db/db_val.h */ VAL_TYPE(vals) = DB_STR; VAL_NULL(vals) = 0; VAL_STR(vals).s = _host->s; VAL_STR(vals).len = _host->len; /* LM_INFO(">>domain.c>>-1-connector=%d and tab=%s\n",db_handle,trusdomain_table.s); */ if (domain_dbf.use_table(db_handle, trusdomain_table.s) < 0) { LM_ERR("Error while trying to use trusdomain table\n"); return -1; } if (domain_dbf.query(db_handle, keys, NULL, vals, cols, 1, 1, 0, &res) < 0) { LM_ERR("Error while querying database\n"); domain_dbf.close(db_handle); return -1; } /* RES_ROW_N(res) Num. de linhas no fetch */ if (RES_ROW_N(res) <= 0 || RES_ROWS(res)[0].values[0].nul != 0){ LM_DBG("no value found\n"); if (res != NULL && domain_dbf.free_result(db_handle, res) < 0) LM_DBG("failed to free the result\n"); domain_dbf.close(db_handle); } else if (RES_ROW_N(res) >= 1) { /* Teste do que estah na tab. trusdomain com o que vem na URI */ LM_DBG("Realm '%.*s' is trusted\n",_host->len, ZSW(_host->s)); } /* Free the result */ domain_dbf.free_result(db_handle, res); domain_dbf.close(db_handle); return 0; } *And declaration in domain_mod.c :* ... /* by mrg */ #define TRUSDOMAIN_TABLE "trusdomain" #define TRUSDOMAIN_TABLE_LEN (sizeof(TRUSDOMAIN_TABLE) - 1) #define TRUSDOMAIN_COL "trusdomain" #define TRUSDOMAIN_COL_LEN (sizeof(TRUSDOMAIN_COL) - 1) ... ... str trusdomain_table = {TRUSDOMAIN_TABLE, TRUSDOMAIN_TABLE_LEN}; /* Name of trusted table */ str trusdomain_col = {TRUSDOMAIN_COL, TRUSDOMAIN_COL_LEN}; /* Name of trusted column */ ... ... /* * * Exported functions */ static cmd_export_t cmds[] = { {"is_from_local", is_from_local, 0, 0, 0, REQUEST_ROUTE|BRANCH_ROUTE|FAILURE_ROUTE}, {"is_uri_host_local", is_uri_host_local, 0, 0, 0, REQUEST_ROUTE|BRANCH_ROUTE|FAILURE_ROUTE}, {"is_domain_local", w_is_domain_local, 1, parameter_fixup, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {"is_from_trusted", is_from_trusted, 0, 0, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {0, 0, 0, 0, 0, 0} }; .../* * Exported parameters */ static param_export_t params[] = { {"db_url", STR_PARAM, &db_url.s }, {"db_mode", INT_PARAM, &db_mode }, {"domain_table", STR_PARAM, &domain_table.s}, {"domain_col", STR_PARAM, &domain_col.s }, {"trusdomain_table", STR_PARAM, &trusdomain_table.s}, {"trusdomain_col", STR_PARAM, &trusdomain_col.s}, {0, 0, 0} }; ... ... static int mod_init(void) { int vertr....; ..... trusdomain_table.len = strlen(trusdomain_table.s); trusdomain_col.len = strlen(trusdomain_col.s); ..... if (db_mode != 0) { if (domain_db_init(db_url.s)<0) return -1; /* Check table version */ ver = domain_db_ver(&domain_table); vertr = domain_db_ver(&trusdomain_table); if ((ver < 0) && (vertr < 0)) { LM_ERR("Error while querying table version\n"); goto error; } else if ((ver < TABLE_VERSION) && (vertr < TABLE_VERSION)) { LM_ERR("Invalid table version <%d> trustable version <%d> (should be %d)\n", ver, vertr, TABLE_VERSION); goto error; } ....... ....... /* Initializing hash tables and hash table variable */ ........ domain_db_close(); } return 0; error: domain_db_close(); return -1; } So I think I must be doing everything wrong ..... as would be the right and as I could learn so fast as this error debugging of pID of Openser is dead...??? Any help would be greatly appreciated... thx Marcio

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Re: [Kamailio-Users] mediaproxy on centos 5.x
by Ricky Gutierrez 13 Sep '08

13 Sep '08

Hi list, this was the problem my version of libnetfilter-conntrack-dev thank Dan and list rickygm ----- Original Message ---- From: Dan Pascu <dan(a)ag-projects.com> To: Ricky Gutierrez <xserverlinux(a)yahoo.com> Sent: Friday, September 12, 2008 4:57:14 AM Subject: Re: mediaproxy on centos 5.x On Friday 12 September 2008, Ricky Gutierrez wrote: > Hi Dan , I am trying to install mediaproxy on a centos 5.2, but when I > am Settling it shows this error and it doesn't finish installing > > mediaproxy/interfaces/system/_conntrack.c: In function > âForwardingRule_initâ: mediaproxy/interfaces/system/_conntrack.c:223: > error: âNFCT_SOPT_SETUP_REPLYâ undeclared (first use in this function) > mediaproxy/interfaces/system/_conntrack.c:223: error: (Each undeclared > identifier is reported only once > mediaproxy/interfaces/system/_conntrack.c:223: error: for each function > it appears in.) error: command 'gcc' failed with exit status 1 > > I have followed this mail, I have installed these dependences but it > even fails me Make sure you have libnetfilter-conntrack-dev at least version 0.0.89 The symbol it doesn't find was introduced on 4 Jun 2007, so I guess you have a pretty old version (it first appeared in 0.0.80, but we recommend you use at least 0.0.89 since that is what we used while developing it and it's guaranteed to work with). Better yet use the latest version (0.0.96) > > http://lists.opensips.org/pipermail/users/2008-August/000131.html > > could you help me? > > God blesses you > > rickygm -- Dan

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[Serusers] ser authentication user
by Amos Shahar 12 Sep '08

12 Sep '08

Hi, how can I add user to ser where the auth user is diffrent the user (e.g my DID)? for example I have: user 12127773456 (my number in the header, from, contact....) pass 1234 but the authentication user is 10000 (for the autherization proxy) using serctl does not have this parameter: serctl add <username> <password> <email> .. add a new subscriber for example in X-lite I want to have diffrent user name and autherization user name and to be able to register to SER. Thanks! Amos

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[Kamailio-Users] SIP NOTIFY for Message-Waiting
by George Lee 12 Sep '08

12 Sep '08

Hi, I know Kamailio SIP server can handle SUBSCRIBE for message-waiting indication. Is there a way to ask the SIP server to generate a NOTIFY with message summary event and its related headers like Message-waiting, Message-Account, Voice-Message, etc.? Or I must use Asterisk as my mail server and forward all mwi Subscribes to Asterisk? Thanks, George

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[Kamailio-Users] after an append_hf
by BERGANZ François 12 Sep '08

12 Sep '08

hello all I need help for 'append_hf' : I know how to add sip header with append_hf but how can look at this after (in another server) ? I need to do append_hf("user: XXX"\r\n"); and after (with another OpenSer) look at the string after user: ? How to do? Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

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[Kamailio-Users] Changing Source IP per Route
by Ali Jawad 12 Sep '08

12 Sep '08

Hi All Can the following be achieved.: I do have a subscription at a carrier with two routes, on normal route and one premium route. The problem is that I do have to use a different source IP with each of these two routes. This would be easy to achieve with two different openser boxes. However I do have only one at current time. So, is it possible to use a different source IP per route ? Thanks With Regards Ali Jawad

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[Kamailio-Users] branch_route and onreply_route for CANCEL and ACK
by Alex Hermann 12 Sep '08

12 Sep '08

Hello, I use t_onbranch and t_on_reply for all request passing the proxy. Why doesn't the branch_route get called for CANCEL's and ACK's (loose routed as well as statefull)? The loose routed BYE does pass through the branch_route. And why isn't the reply on a CANCEL passed through the transactional onreply_route? It is passed through the default onreply_route. The documentation doesn't mention any selectivity on request method. -- Greetings, Alex Hermann

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[Serusers] DNS server and registered users
by Magnus Sörman 12 Sep '08

12 Sep '08

Hi, How do I do the following in SER 2.0: 1. How do I specify which DNS server that should be used? E.g. That a reverse look-up should go to the DNS server running on the same machine as SER, instead of the once specified in /etc/resolv.conf. 2. If I'm not using the mysql database, how can I see how many users that are registered in usrloc/RAM? Btw, 'ser_ctl user show <uri>' didn't even find the successfully registered users, so maybe that's something I somehow need to address first. Thanks, /Magnus

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[Serusers] How to use record route to change path of BYE messages
by 邱磊 12 Sep '08

12 Sep '08

Hi,everyone For voice communication I use a SIP+RTP proxy together with SER. For PSTN calls SER routes the INVITE messages to the RTP proxy. Everything works fine there, but when the call is ended by one client (which is connected to SER) or the PSTN user, the BYE message arrive to SER but dont forward it. If our RTP proxy does not see the message coming from SER, call doesn't end. That's why I need all of the BYE messages to go through SER. How can I do that ? In the documents they said that the bye message loose route if there are ";lr" and If the message should be loose routed then SER should simply relay the message to the next destination as specified in the top-most Record-Route header field .but i have a try and nothing happened! that is my config block: # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" # ------------------------- request routing logic ------------------- # main routing logic route { record_route(); if(message=="BYE") loose_route(); }; my ser is 1.2.3.4:5060 this is my sip message: BYE sip:0000123@19.68.150.20:31562 SIP/2.0^M Via: SIP/2.0/UDP 23.43.53.4:5060;branch=z9hG4bK5a69ae91;rport^M Record-Route: <sip:1.2.3.4:5060;ftag=28118cbe-30554463;lr>^M Record-Route: <sip:12.45.32.61:5060;r2=on;lr;ftag=28118cbe-30554463>^M From: <sip:123456@1.2.3.4:5060;user=phone>;tag=as7232fbd6^M To: "beurself212" <sip:0000123@19.68.150.20:31562>;tag=28118cbe-30554463^M Call-ID: 76ad6e55-86f19c97-MzA1NTQ0NjM(a)192.168.50.50^M Thanks, Jimmy 2008-09-12 邱磊

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[Serusers] failure route problem - duplicate invites
by Matthew McGuire 11 Sep '08

11 Sep '08

I'm having a problem with the route sections below. When SER receives an Invite to proxy to a PSTN gateway, SER is sending 2 invites very fast to the gateway in the first route block which ends up creating a mess with the gateway. Either I have something wrong in the routing code or I have an SER/OS/hardware timing problem. Any suggestions ? route[1] { xlog("L_INFO", "Sending to route 1\n%mb\n"); rewritehostport ("1.1.1.1:5060"); append_branch(); t_on_failure("1"); t_relay(); } failure_route[1] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 1\n%mb\n"); append_branch(); route(2); break; } route[2] { xlog("L_INFO", "Sending to route 2\n%mb\n"); rewritehostport ("2.2.2.2:5060"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 2\n%mb\n"); append_branch(); route(3); break; } route[3] { xlog("L_INFO", "Sending to rout 3\n%mb\n"); rewritehostport ("3.3.3.3:5060"); t_relay(); } Thanks, Matt

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Re: [Kamailio-Users] help installing mediaproxy in centos 5
by Ricky Gutierrez 11 Sep '08

11 Sep '08

I install the dependences in my centos "kernerl-devel" and the netfilter kernel module . "libnetfilter-conntrack-dev" but it still shows me the installation error I read this mail, thinking that that it could be my problem http://lists.opensips.org/pipermail/users/2008-August/000131.html thank you for all help that I can be offered rickygm ----- Original Message ---- From: Peter Lemenkov <lemenkov(a)gmail.com> Sent: Wednesday, September 10, 2008 11:13:57 PM Subject: Re: [Kamailio-Users] help installing mediaproxy in centos 5 > I've got different advise. After quick googling I found that these >constants are related to netfilter kernel module, so you probably need > to install kernel-headers and pass properly additional includedirs (if > required) to mediaproxy buildsystem.

1 0

[Serusers] Difficulties "making" SER v2
by Steven C. Blair 11 Sep '08

11 Sep '08

I'm trying build make SERv2 on a RedHat 2.6.18-92.1.10.el5xen SMP machine. I'm having difficulty with include definition conflicts. Are there specific instructions for making SER on 64bit architecture machines? The first error occurs with futex.h Thanks

1 0

[Kamailio-Users] custom module depends on usrloc, the load order
by toly 11 Sep '08

11 Sep '08

Greetings, I have 2 questions. 1. I've created custom module where I'd like to get number of users from usrloc module. Here is the code: LOG(L_INFO,"PRESOURCE: Initializing\n"); bind_usrloc = (bind_usrloc_t)find_export("ul_bind_usrloc", 1, 0); if ( !bind_usrloc ) { LOG(L_CRIT, "Can't bind usrloc (find_export) check if loaded"); goto error; } if ( !bind_usrloc(&ul) ) { LOG(L_CRIT, "Can't bind usrloc"); goto error; } When this code is enabled if fails on second if. Looks like usrloc module is not loaded. Here's the log 09/10 18:50:57 4431 info <statistics.c: 135> INFO: statistics manager successfully initialized 09/10 18:50:57 4431 info <uac.c: 139> UAC - initializing 09/10 18:50:57 4431 info <hslot.c: 44> INFO:ul_init_locks: locks array size 512 09/10 18:50:57 4431 info <pike.c: 108> PIKE - initializing 09/10 18:50:57 4431 info <ip_tree.c: 86> INFO:pike:init_lock_set: probing 256 set size 09/10 18:50:57 4431 info <avpops.c: 164> AVPops - initializing 09/10 18:50:57 4431 info <sl.c: 130> StateLess module - initializing 09/10 18:50:57 4431 info <tm.c: 586> TM - initializing... 09/10 18:50:57 4431 notice <maxfwd.c: 97> Maxfwd module- initializing 09/10 18:50:57 4431 info <presource.c: 146> PRESOURCE: Initializing 09/10 18:50:57 4431 crit <presource.c: 155> Can't bind usrloc 09/10 18:50:57 4431 err <sr_module.c: 465> init_mod(): Error while initializing module presource In config script loadmodule("usrloc.so") is prior to load my custom module. When I disable above code, then load order follows config file order. 2. Here's how I wanted to get number of users: unsigned long get_number_of_users() { return ul.get_number_of_users(); } But usrloc_api_t does not contain this cmd function despite it's exported in usrloc module typedef struct usrloc_api { int use_domain; int db_mode; unsigned int nat_flag; register_udomain_t register_udomain; get_all_ucontacts_t get_all_ucontacts; insert_urecord_t insert_urecord; delete_urecord_t delete_urecord; get_urecord_t get_urecord; lock_udomain_t lock_udomain; unlock_udomain_t unlock_udomain; release_urecord_t release_urecord; insert_ucontact_t insert_ucontact; delete_ucontact_t delete_ucontact; get_ucontact_t get_ucontact; update_ucontact_t update_ucontact; register_watcher_t register_watcher; unregister_watcher_t unregister_watcher; register_ulcb_t register_ulcb; } usrloc_api_t; Thanks, Toly -- View this message in context: http://www.nabble.com/custom-module-depends-on-usrloc%2C-the-load-order-tp1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] Sarfaraz has invited you to join iDeezire - Keeps Connected!
by Sarfaraz 11 Sep '08

11 Sep '08

1 0

[Serusers] Sarfaraz has invited you to join iDeezire - Keeps Connected!
by Sarfaraz 11 Sep '08

11 Sep '08

1 0

[Kamailio-Users] preparing 1.4.1 release
by Daniel-Constantin Mierla 11 Sep '08

11 Sep '08

Hello, there are already several important fixes since 1.4.0, so we should look to build the first maintenance release for branch 1.4. I am thinking of 2 weeks time frame from now. Make sure you point outstanding issues you are aware of as soon as possible on the tracker. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] help installing mediaproxy in centos 5
by Ricky Gutierrez 11 Sep '08

11 Sep '08

Hi list is trying to install mediaproxy 2.0.3 in centos 5 and when I make python ./setup.py install it generates me this error: mediaproxy/interfaces/system/_conntrack.c:490: warning: implicit declaration of function ânfct_catchâ mediaproxy/interfaces/system/_conntrack.c:491: warning: implicit declaration of function ânfct_callback_unregisterâ mediaproxy/interfaces/system/_conntrack.c:502: warning: implicit declaration of function ânfct_get_attr_u8â mediaproxy/interfaces/system/_conntrack.c:502: error: âATTR_ORIG_L3PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:503: error: âATTR_REPL_L3PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:504: error: âATTR_ORIG_L4PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:505: error: âATTR_REPL_L4PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:513: error: âATTR_ORIG_IPV4_SRCâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:514: error: âATTR_ORIG_PORT_SRCâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:515: error: âATTR_DNAT_IPV4â undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:515: error: âATTR_REPL_IPV4_SRCâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:516: error: âATTR_DNAT_PORTâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:516: error: âATTR_REPL_PORT_SRCâ undeclared (first use in this function) error: command 'gcc' failed with exit status 1 I already have installed the bookstore python any suggestion is appreciated? best regards rickygm

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[OpenSER-Users] Problems with SerMyAdmin Installation
by Mark Koopmann 10 Sep '08

10 Sep '08

Hi everybody, does anybody has experience with installing the java tool SerMyAdmin? The Tool is running but every action ends with an runtime exception. Trying to register a new user ends with the following message: Grails Runtime Exception Error Details Message: Caused by: java.lang.NullPointerException Class: RegisterUserController At Line: [98] Code Snippet: (empty) Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: java.lang.NullPointerException at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595)Caused by: java.lang.NullPointerException at RegisterUser.toString(RegisterUser.groovy) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy:98) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy) ... 29 more Unfortunately, i dont have much experience with installing such java stuff. It would be great if one of you can give me a hint or something. I dont have any idea how to solve this problem. Thanks for any comment! Best regards Mark Koopmann

5 7

[Kamailio-Users] controle the dispatcher
by BERGANZ François 10 Sep '08

10 Sep '08

Hello, I need to check where is send the frame to the next server, I need to do something like: ds_select_dst("1","0"); xlog("L_ERR", " methode <$rm> ruri:$ruri ru:$ru\n"); forward(); exit; I want to log to next server! How to do? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

3 2

[Kamailio-Users] Prefix with variable?
by Robin Vleij 10 Sep '08

10 Sep '08

Hi! I'm running into a little problem using the prefix() function. According to the docs I can only use strings as prefixes and this works fine everywhere in our config. Now I'm trying to do something like this: prefix($avp(s:t_prefix)); Customer should be able to choose their own prefixes for certain types of traffic. We get a custom prefix in from the PSTN gateways, where we want to then replace that "internal" prefix (strip(4)) with a configurable prefix. Why can't we use variables in the prefix function? I could try to use $ru = $avp(s:t_prefix) + $original_ru or something, but on a short try-run this didn't seem as easy as that. Any ideas or anyone doing something like this? /robin -- Robin Vleij robin(a)swip.net

2 2

[Kamailio-Users] CANCEL can't be forward!
by BERGANZ François 10 Sep '08

10 Sep '08

hello a problem with my CANCEL frames! user-------openser------asterisk invite>>>>>>>|>>>>>>invite ringing<<<<<<|<<<<<<ringing cancel>>>>>>>| in fact, I don't know where forward the cancel in stateless! I have no information in the frame to forward it to the right asterisk (one for g711, one for g729...) have you an idea? thank you all Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

3 4

[Kamailio-Users] failure route problem - duplicate invites
by Matthew McGuire 10 Sep '08

10 Sep '08

I'm having a problem with the route sections below. When my proxy receives an Invite to proxy to a PSTN gateway, the proxy is sending 2 invites very fast to the gateway in the first route block which ends up creating a mess with the gateway. Either I have something wrong in the routing code or I have an SER/OS/hardware timing problem. Any suggestions ? route[1] { xlog("L_INFO", "Sending to route 1\n%mb\n"); rewritehostport ("1.1.1.1:5060"); append_branch(); t_on_failure("1"); t_relay(); } failure_route[1] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 1\n%mb\n"); append_branch(); route(2); break; } route[2] { xlog("L_INFO", "Sending to route 2\n%mb\n"); rewritehostport ("2.2.2.2:5060"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 2\n%mb\n"); append_branch(); route(3); break; } route[3] { xlog("L_INFO", "Sending to rout 3\n%mb\n"); rewritehostport ("3.3.3.3:5060"); t_relay(); } Thanks, Matt

5 11

[Serusers] help required
by Engr. Muhammad Noman Saeed 10 Sep '08

10 Sep '08

Dear All, I need your help for establishing SER as a SIP server for my organization. I have downloaded it at my fedora 8.0 machine, but as i am new with linux and SER it is hard to find me how to install/configure. kindly help me as a new SER user. regards -- ========================= Noman =========================

2 1

[Serusers] how can i config ser.cfg
by 邱磊 10 Sep '08

10 Sep '08

hi friends: i need some help! frist,my asterisk receive a sip message from sip proxy via SER.when my asterisk send 200 to SER and then receive ACK from proxy directly! when asterisk send BYE to the proxy and can hangup the call. but now i redirect the BYE to SER, i found the SER can not send the message to proxy! what can i do in the ser.cfg? thanks in advance 2008-09-10 邱磊

1 0

[Kamailio-Users] add number to an avp ?
by BERGANZ François 09 Sep '08

09 Sep '08

I need to add 1 to an avp like : $avp(i:681)=$avp(i:681)+1; but don't work! how to? thank you Cordialement BERGANZ François http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

2 2

[Serusers] Dubt on RFC 4484 and SIP Proxy Server behavior
by Francesco la Torre 09 Sep '08

09 Sep '08

Hi all, I've a doubt regarding a statement at the end of page 5 "... Some important design decisions are associated with carrying assertions in a SIP request. If an assertion is carried by value, or uses a MIME-based content indirection system, then proxy servers will be unable to inspect the assertion themselves. If the assertion were referenced in a header, however, it might be possible for the proxy to acquire and inspect the assertion itself. There ..." Why a proxy server will be unable to inspect the assertion themself ? SER has functions by which it could be read message body, does this behavior not respect any rfc ? I'm missing something :-S Thanks in advance, Francesco la Torre

2 4

[Kamailio-Users] [Fwd: Re: Unconditional call forward]
by Stefano Palleschi 09 Sep '08

09 Sep '08

Hi Gustavo, thanks for your reply. I've changed the *avp type* from "2" to "0" but I get always the same error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> Now I'm going to reboot the server... If I'll have news I'll let you know. Regards, Stefano -------- Messaggio Originale -------- Oggetto: Re: [Kamailio-Users] Unconditional call forward Data: Wed, 3 Sep 2008 10:38:42 -0300 Da: Gustavo Mistrinelli <gmistrinelli(a)gmail.com> A: Stefano Palleschi <stefano.palleschi(a)okcom.it> CC: users(a)lists.kamailio.org Referenze: <48BE6299.5000805(a)okcom.it> Hi Stefano, Please check *avp type* that you are using, use type 0 instead of 2 See below the different types for avps: 0 - *AVP* with string name and string value 1 - *AVP* with string name and integer value 2 - *AVP* with integer name and string value 3 - *AVP* with integer name and integer value Regards, Gustavo On Wed, Sep 3, 2008 at 7:10 AM, Stefano Palleschi <stefano.palleschi(a)okcom.it <mailto:stefano.palleschi@okcom.it>> wrote: Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 <http://81.29.176.57> | callfwd | sip:0666653218@81.29.176.57 <mailto:sip%3A0666653218@81.29.176.57> | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57 <http://81.29.176.57>> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi _______________________________________________ Users mailing list Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- Gustavo Mistrinelli

2 2

[Kamailio-Users] 302 redirect inside redirect
by Kent Pirlo 09 Sep '08

09 Sep '08

Ok, I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch. call flow would be something like this: call comes into kamailio, 1) kamailio sends the invite to another proxy(1), 2) that proxy returns a 302,(if it times out it moves on to step 5) 3) kamailio sends an ACK back to that proxy(1), 4) kamailio looks at the contact header of the 302 reply, 5) depending on that info, I decide where to tell the originating switch where to go, 5) kamailio sends originating switch a 300 redirect with new contact header. 6) originating switch sends back ACK. I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy. The onreply route, graps the contact saves it into variable and then sends to route(2), Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect"). The switch then sends the ACK, but kamailio says its not a local ACK and ignores it.. Then it keeps sending 300's back to the switch... Please help.. Thanks in Advance. kent

2 1

[Kamailio-Users] Documentation Site Down
by Jon Farmer 09 Sep '08

09 Sep '08

Hi When I go to http://kamailio.org/docs/ i get a 404 error. Any ideas anyone? Regards Jon

5 5

[Kamailio-Users] Forward call
by Thiago Rondon 09 Sep '08

09 Sep '08

Folks, My problem for forward call, that I need to change username/password and realm to do my forward. So, I try this: modparam("uac","from_restore_mode", "auto") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") modparam("uac","auth_username_avp","$avp(s:uac_username)") modparam("uac","auth_password_avp","$avp(s:uac_password)" route { ... if(uri=~"^sip:0[0-9]{10}@") { xlog("L_INFO","uac_atual: $avp(s:uac_username) !!"); $avp(s:uac_username)="username"; $avp(s:uac_realm)="1.1.1.1"; $avp(s:uac_password)="password"; xlog("L_INFO","uac_trocado $avp(s:uac_username) !!"); route(3); exit; } .. } ... route[3] { t_on_failure("3"); # reset flag to mark no authentication yet performed resetflag(7); # forward to PSTN uac_replace_from("username","sip:username@1.1.1.1"); rewritehostport("1.1.1.1:5060"); xlog("L_INFO","rewritehost !!!!!!!!"); t_relay(); } .. So, when I do xlog, in the first entry, uac_username don't display, why ? And its strange, because in one supplier its ok, and in another its not run. But I don't undestand why uac_username is empty in the first xlog! Thanks, -Thiago Rondon

2 1

[Kamailio-Users] populate_leg_info
by David Villasmil 08 Sep '08

08 Sep '08

Anyone lnow what this means? dialog:populate_leg_info: bad sip message or missing Contact hdr ERROR:dialog:dlg_onreq: could not add further info to the dialog openser 1.3.2 thnks

2 1

[Kamailio-Users] New OpenXCAP release 1.0.0
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello, There is a new release of OpenXCAP server available. It contains many bug fixes and additions for important features like XML partial manipulation, xcap-caps, xcap-diff and RLS services that were not available at the initial release last year. With this features OpenXCAP fulfills now most if not all relevant requirements for an XCAP server implementation. A generic xcap client library and command line client is available in the clients directory, you can use it to manipulate documents on OpenXCAP or other XCAP server implementation. The server software is also operational at http://sip2sip.info so you can test against it by registering a SIP account. Changes in version 1.0.0 * Added RLS services (RFC 4662 and RFC 4826) * Added support for xcap-diff based on draft-ietf-simple-xcap-diff-09 * Added partial get/put/delete of elements in the XCAP documents * Added test suite for rls-services, resource-lists and partial updates * Many bug fixes from field experiences * Development status from beta to production * Switched to Python 2.5 * Improved documentation and testing suite * Changed license to GPL * Fixed MySQL operational error (2006) The software can be downloaded from: http://openxcap.org Many thanks to Denis Bilenko who did most of the works for this new release. Kind regards, Adrian Georgescu

1 0

[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello Thanks for the feedback from everyone. You can try cdrtool version 6.6.0, I have added parallel prepaid call support. The algorithm is described in PREPAID.txt cdrtool (6.6.0) unstable; urgency=low * Allow simultaneous calls for prepaid accounts, see the Simultaneous prepaid calls section in doc/PREPAID.txt * Added pseudo code examples for call control engine * Apply the changes from setup/mysql/alter_tables.mysql * Added interface to kill ongoing sessions in Prepaid page Testing and feedback is useful and welcome. Regards, Adrian

1 0

[Kamailio-Users] call forward problem
by muhammad akl 06 Sep '08

06 Sep '08

hey all after making call forward feature using avpops module and its exported functions avp_db_load and avp_pushto i have a problem : assuming all calls to extension 12001 will be forwarded to extension 12008, in opensers's database i have added the following : username=12001 domain =mydomain attribute = callfwd value = sip:12008@mydomain type = 0 untill that every thing was fine and all calls comes to 12001 forward to 12008 without any problem , the problem comes from when i am using 12008 to call 12001 it calls itself ( mean 12008) so anyone can tell me what i have missed ? i have configured my openser.cfg like this in loaded modules section added loadmodules "avpops.so" > in module parameter section added : modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") in Invite message handler added : if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } any suggestions ?

4 7

[Kamailio-Users] two proxies sharing the same database
by toly 05 Sep '08

05 Sep '08

Greetings, I have 2 proxies having the same database on backend. One is the primary, second is standby - OpenSER 1.2.1. That's how I implement DNS based failover. What I see, when all UA's registered on primary and I start standby, I's starting to cleanup location table according to it's own timer. So record is gone, when primary trying to updated the record on re-register it of course fails, as a result location table is wrong. What will be a solution? Thanks, Toly -- View this message in context: http://www.nabble.com/two-proxies-sharing-the-same-database-tp19335185p1933… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Kamailio-Users] How to solve early media during serial forking
by Iñaki Baz Castillo 05 Sep '08

05 Sep '08

Hi, AFAIK there is an issue in typical SIP/PSTN enviroments in the case of failover using different gateways serially. The main issue is that SIP doesn't define how a UAC should handle the case in which it receives various "183 Session Progress" from different early-dialogs, each of them containing early-media. It occurs to me that a SIP/PSTN gateway sometimes replies 183 with early media for 1 or 2 seconds, and inmediatelly it replies a 502. Well, this 502 could be caused by other node in the PSTN network, it's not necessarialy a problem fo the gateway [1]. In this case I do serial forking forwarding the request to other gateway. But the UAC already received early media from the first gateway, and most of the SIP phones just render the first early media they receive in case of multiple 183, and drop others. So basically the problem is that the UAC user doesn't listen the ringing coming from PSTN after the request has been forwarded to the second gateway. Is there something I could do at proxy level to avoid this issue? Sincerely I don't think so, since it makes no sense to disallow the first 183 (I cannot know that the gateway will later reply a 502). Thanks. [1] Anyway, *who* is creating the early media? Why the PSTN generates it if the request doesn't arrive to the called? "imagine there is no PSTN..." -- Iñaki Baz Castillo

1 0

[Kamailio-Users] custom routing
by toly 05 Sep '08

05 Sep '08

Greetings, Suppose I need to do custom routing for outbound calls from UA - softphone. Each user account may have the following info attached to it: 1:provider_a:2:provider_b:3:provider_c which means: forward call to provider_a, if failed forward call to provider_b if failed forward call to provider_c if failed send response back to UA I do database lookup based on from, and get above string. Suppose I wrote my own dispatcher module which initially gets set to above string: 1:provider_a:2:provider_b:3:provider_c and forwards to first provider, then on failure_route I call next_provider and so on. Will it work across multiple messages? Will it work correctly for each child - I mean joe@domain and paul@domain should be handled according their own providers? Any other suggestions? Thanks, Toly -- View this message in context: http://www.nabble.com/custom-routing-tp19322478p19322478.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 4

[Kamailio-Users] connection to remote mysql database
by Asim Riaz 05 Sep '08

05 Sep '08

I am using Kamailio 1.2 , in t openser.cfg I defined the the url for db, which is not local ("auth_db|permissions|uri_db|usr_loc", "db_url", "mysql://openser:db_password@db_ip_address/openser") but still getting the error message below when start openser; /sbin/openser[13064]: permissions - initializing /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.allow /sbin/openser[13064]: Default allow file (//etc/openser/permissions.allow) not found => empty rule set /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.deny /sbin/openser[13064]: Default deny file (//etc/openser/permissions.deny) not found => empty rule set /sbin/openser[13064]: new_connection: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) /sbin/openser[13064]: ERROR: group_db_bind: unable to connect to the database /sbin/openser[13064]: ERROR:group:mod_init: unable to open database connection /sbin/openser[13064]: init_mod(): Error while initializing module group /sbin/openser[13064]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill any idea ? thanks in advance Asim

2 2

[Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you

2 3

Re: [Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

I could have a look, but Asterisk-Openseràno problems Centile-openseràdont work ! Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15 De : Darren Sessions [mailto:dmsessions@gmail.com] Envoyé : vendredi 5 septembre 2008 15:26 À : BERGANZ François Objet : Re: [Kamailio-Users] Openser & Centile Would be helpful to see some of the SIP signaling. _____________________________ Darren Sessions dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Sep 5, 2008, at 7:24 AM, BERGANZ François wrote: Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 05 Sep '08

05 Sep '08

The problem with concurrent prepaid calls and single balance is that you have to correlate between the call control and rating angine somehow so that all calls terminate when balnce becomes zero. The problem is a bit complex: Example: Balance = 10. A call starts to destination XXX, for the sake of example max session time = 2 minutes After one minute, you start second call to destination YYY which has a different price and your balance is not anymore 10 but depends on the duration of the first call which is in progress. What is the maximum session time for it given that the first call is already in progress? What should happen with the first call? I am looking for suggestions on implementing a proper algorithm to deal with this situation in the rating engine. If you have any I would be glad to hear it. Adrian

7 8

[Serusers] /tmp/ser_fifo disaapears after rebooting
by Joseph Myalla 05 Sep '08

05 Sep '08

I have revmoved the ser and reinstalled again, just after the installation ser works, i can see the PID, calls are made and text messages between computers, lokking in the tmp/ser_fifo is availabe, but it desappears after rebooting, what could be the reason On Wed, Sep 3, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. Error opening ser's fifo /tmp/ser_fifo (Joseph Myalla) > 2. Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu > 8.04LTs (Joseph Myalla) > 3. Re: Error opening ser's fifo /tmp/ser_fifo (Atle Samuelsen) > 4. SER 0.9.7-pre3 and timers (Steven C. Blair) > 5. how can i get the ip ? ( ?? ) > 6. how can i get the ip? ( ?? ) > 7. how can i get the IP? ( ?? ) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 2 Sep 2008 12:12:39 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020412n27b87d6g8f7346a569d98286(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 6.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 2 > Date: Tue, 2 Sep 2008 12:14:24 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry > its ubuntu 8.04LTs > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020414q25b69c15ya029023b51ce367e(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 8.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 3 > Date: Tue, 2 Sep 2008 13:18:02 +0200 > From: Atle Samuelsen <clona(a)cyberhouse.no> > Subject: Re: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: Joseph Myalla <chanangae(a)gmail.com> > Cc: serusers(a)lists.iptel.org > Message-ID: <20080902111802.GD63335(a)mystic.cyberhouse.no> > Content-Type: text/plain; charset=us-ascii > > Hi Joseph, > > have you started ser? because, if ser is not running you will not be > able to get access to /tmp/ser_fifo. > > You can also check if it's there by doing a ls /tmp/ser_fifo :) > > > - Atle > > > * Joseph Myalla <chanangae(a)gmail.com> [080902 13:13]: >> I have downloades ser's debian packages and installed them in Ubuntu >> Linux 6.0Ls Desktop edition >> ser_0.9.7_i386.deb and the correspoding ser_mysql modules, >> >> trying commands serctrl ps or ul show i get an error "Error opening >> ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I >> get error SER/FIFO not accessible: 2 >> >> the are some instructions in the mailing list saying I should add >> fifo_mode=0777 in the configuration file, I have done it but still the >> errors exist. >> >> can any one help. >> >> >> -- >> Myalla J.C >> Kibaha Education Centre >> P.O Box 30054 Kibaha >> Pwani >> Tanzania >> +255787680744 >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > ------------------------------ > > Message: 4 > Date: Tue, 2 Sep 2008 11:14:46 -0400 > From: "Steven C. Blair" <blairs(a)isc.upenn.edu> > Subject: [Serusers] SER 0.9.7-pre3 and timers > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: > <20C52563FC353043A5A9B80D847590E102D6824900(a)MAILBOX.exchange.upenn.edu> > > Content-Type: text/plain; charset="us-ascii" > > > I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. > > Thanks, Steve > > > > >

2 1

[Kamailio-Users] Mediaproxy vs. RTPproxy.
by Alex Balashov 05 Sep '08

05 Sep '08

This must be an incredibly dumb question, but why would anyone ever use mediaproxy over rtpproxy? If one is written in Python and one in C, it seems to me that the performance and scalability makes the choice of rtpproxy a no-brainer. What benefit is there to using mediaproxy? More features and capabilities, ease of configuration perhaps? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 6

[Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
by Ali Jawad 05 Sep '08

05 Sep '08

Hi All I am using using openser 1.3..if I make a call between two softphones on the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode and the call get's hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the logs of openser I can't see any errors. However on the wire shark I can see icmp destination unreachable...port unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri: <sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200 8F980980E97E42F8EC;nat=yes> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232, <branch> = Shouldn't the transport=TLS ? Regards

4 15

[Serusers] how can i get the ip?
by 邱磊 05 Sep '08

05 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regards 2008-09-05 邱磊

1 0

[Kamailio-Users] Help with SerMyAdmin
by Tiago Rés 04 Sep '08

04 Sep '08

Hello all, I'm a new user in this list, but I have already seen some posts about serMyAdmin. I have openser 1.2.2 installed, running with mysql Version 14.12. I'm trying to use a web interface for openser, such as sermyadmin or openser administrator, but I'm unable to accomplish that. I have followed the instructions from sermyadmin website, but I had some errors: I couldn't add a user in mysql with that command, so I used something like: openserctl add admin openserrw admin@localhost The remaining steps went ok, but everytime I try to open http://(myserver):8080/serMyAdmin I get an error page from Tomcat. Any suggestion? Thanks in advance. Tiago

2 3

[Kamailio-Users] Fax Detection
by Abdul Hakeem 04 Sep '08

04 Sep '08

Hello, Is there a way to identify in incoming call as a fax (perhaps by detecting CNG tone ). Regards, Abdul Hakeem

2 1

[Kamailio-Users] ERROR:mysql:db_mysql_new_connection: driver error: Too many connections
by Samael - 04 Sep '08

04 Sep '08

Hi, We are deploying OpenSER in a large infrastructure together with Asterisk to support thousands of current transactions. We are using udp listener and mysql module in openser. We had to increase the children parameter of the openser.cfg to 200 because Openser couldn't handle the high amount of concurrent SIP REQUESTs from ASTERISK (Not only INVITES, but also a lot of SIP OPTIONs), that means that Openser couldn't handle more than 'children' REQUEST loosing the rest of the concurrent REQUEST. children=200 But then, the following error appeared in the openser: ERROR:mysql:db_mysql_new_connection: driver error: Too many connections This is because the Standard parameter of mysql "max_connections" is set to 100. In mysql CLI, we can see that every child process of openser has a mysql connection. (see in Mysql> SHOW PROCESSLIST) The recommended max_connections parameter is 500-1000 => we have a limit of 500-1000 concurrent SIP REQUEST to be processed by the openser. Why there is a process for every child? Why is neccesary a mysql connection for every child? Can the openser have a pool of mysql connection to be used by thousands of children? Regards

4 6

[Kamailio-Users] OpenSER reply 513 when register use FQDN
by Steven Wu 04 Sep '08

04 Sep '08

Hi, I just want use FQDN in my networking environment. My SJPhone use FQDN register to OpenSER 1.2.2 but was rejected by 513 Message to big. However it can be registed use the same profile just replace FQDN to IP address. The DNS query process worked fine and got the right IP address. My environment elments looks like: OpenSER 1.2.2 at host 192.168.1.120 / ipcsip.org DNS server at host 10.57.39.53 SJPhone at host 192.168.1.105 REGISTER package looks like: 192.168.1.105 -> 192.168.1.120 REGISTER sip:ipcsip.org:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062> Contact: <sip:1000@192.168.1.105> Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 192.168.1.120 -> 192.168.1.105 SIP/2.0 513 Message too big Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport=5060 From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.b7d8 Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 I got same result evenif enabled dns & rev_dns in openser.cfg (dns=yes rev_dns=yes). So, What wrong with me? Coudle some give me some instructions? Best regards, Steven Wu

4 7

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 04 Sep '08

04 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Serusers] Which kind of SIP message can I use to exchange Data out of session?
by Peter Cai 04 Sep '08

04 Sep '08

Hi All, If a SIP client has to exchange some data out of a Session with the proxy server, such as get his "buddy list", what SIP Message can I use? I didn't find any thing suit for this purpose. Should I use something beside SIP such as HTTP.

3 3

[Serusers] cannot load module acc.so
by 静于 04 Sep '08

04 Sep '08

hello, I cannot load the acc.so module.The config file is below: # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=10.1.2.3 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=bigu.edu alias=10.1.2.3 # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5060 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. uid="nobody" # group | gid - gid to be used by the server. 99 = nobody. gid="nobody" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external module loading loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ?�?;lr=true?�? modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:>password>@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:>password>@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(10\.1\.2\.3|(proxy\.)?bigu\.edu)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("bigu.edu", "subscriber")) { www_challenge("bigu.edu", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(10\.1\.2\.3|(proxy\.)?bigu\.edu)([:;\?].*)?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { route(3); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@10\.1\.2\.5([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==10.1.2.3) & !(proxy_authorize("bigu.edu", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("bigu.edu", "0"); break; }; }; # to maintain credibility of our proxy, we check >From in INVITEs if (!src_ip==10.1.2.3 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(10.1.2.3|(proxy\.)?bigu\.edu)") { if (!(src_ip==10.1.2.3 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("bigu.edu", "subscriber")) { proxy_challenge( "bigu.edu","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("10.1.2.5:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } Can anyone help BR Ivy ___________________________________________________________ �Ż��䣬��䣡 http://cn.mail.yahoo.com/

1 0

Re: [Kamailio-Users] svn diff
by Vadim Berezniker 04 Sep '08

04 Sep '08

Oops hit Ctrl+Enter svn diff http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/ http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.3/ I've been trying to run this against the public SVN but it never completes.... I hate sourceforge. From: Vadim Berezniker Sent: Wednesday, September 03, 2008 1:57 PM To: 'users(a)lists.kamailio.org' Subject: svn diff Can someone with access to the dev svn run this for me?

3 3

[Kamailio-Users] Could you please help me how to integrate RLS AS and PS AS with SERIMS?
by mahesh.peddi＠info-spectrum.com 04 Sep '08

04 Sep '08

Hi, I am working on openser 1.3.2 as RLS application server and presence application server. I want to integrate this two application servers with SERIMS. Could you please help me how to integrate and make run this. To do this I have run SERIMS components on two servers? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Kristian Kielhofner 03 Sep '08

03 Sep '08

Hello everyone, Feel free to set $OSSPROXY to whatever you like = Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). $OSSPROXY is my favorite SIP toolkit. In addition to being an excellent proxy and registrar it serves as a platform to do SIP mangling and wrangling. With that in mind let me propose a crazy idea... How hard would it be to get $OSSPROXY to be able to parse what is commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has no business getting too involved in the SDP but I feel it makes sense in some instances (like this one). What if I want to make a routing decision based on an encapsulated ISUP parameter of some sort? Who knows what it could be!?!? Maybe there are some other uses but that makes the most sense and seems to be the most practical. It should be possible to develop an $OSSPROXY module to: - Use the SDP parser to find the ISUP SDP part: http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing - Pass that part into a "real" ISUP parser (if necessary). I couldn't seem to find the perfect library but perhaps something like this could work: http://www.openss7.org/isup.html - Make the ISUP parameters available via AVPs, pseudo variables, etc for use in the $OSSPROXY script. I don't really have a need for this but it seems like it could be cool. Am I nuts? -- Kristian Kielhofner http://blog.krisk.org

3 3

[Serusers] Is setdsturi supported in SER
by Steven C. Blair 03 Sep '08

03 Sep '08

I am wondering if setdsturi is supported in SER 0.9.7-pre3 Thanks

1 0

[Kamailio-Users] svn diff
by Vadim Berezniker 03 Sep '08

03 Sep '08

Can someone with access to the dev svn run this for me?

1 0

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 03 Sep '08

03 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Kamailio-Users] VQ_MALLOC memory manager
by Henning Westerholt 03 Sep '08

03 Sep '08

Hi all, does somebody uses the VQ_MALLOC memory manager? I thought about removing this one, as it don't work correctly at the moment (it lacks a realloc implementation). This would be a good cleanup in the core. If somebody still depends on this memory manager, perhaps has also some fixes created for this, please reply to this thread. If nobody speaks up i'll remove the VQ_MALLOC implementation at the end of this month. Cheers, Henning

1 0

[Kamailio-Users] NAT transversal without mediaproxy or rtpproxy
by BERGANZ François 03 Sep '08

03 Sep '08

Hello, I am looking for to do nat transversal without mediaproxy or rtpproxy! I cant find any example without mediaproxy or rtpproxy. Have you an idea? I need to register my users, and contact them after Thank you

4 6

[Kamailio-Users] t_relay to two diffrent proxy
by Asim Riaz 03 Sep '08

03 Sep '08

Hi , I am trying to relay invite from Kamailio to one of two proxies, as long as it receive the reply from one it should not forward to second but incase if the proxy1 does not reply then relay to proxy2. here is my config route(1); if((src_ip != proxy1_ip_address) || (src_ip !=proxy2_ip_address) ) { if(!t_relay_to_udp("proxy1_ip_address", "5060")) || (!t_relay_to_udp("proxy2_ip_address","5060")) { end_media_session(); sl_reply_error(); } } I am also allowing only my proxies can see the location tables; f ((src_ip== Proxy1_ip_address) || (src_ip= Proxy2_ip_address)) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; for some reason incoming calls from my proxies are looping means Kamailio sending back to one of my proxyies, i dont know if there is somthing missing in my configuration or somthing else is causing this problem. Regards Asim Riaz

2 2

[Kamailio-Users] Kamailio dispatcher Module redirection
by Asim Riaz 03 Sep '08

03 Sep '08

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

4 8

[Kamailio-Users] uac_redirect question
by Zahid Mehmood 03 Sep '08

03 Sep '08

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

2 1

[Kamailio-Users] t_relay
by sabino frisardi 03 Sep '08

03 Sep '08

Hi all, I'm using openser 1.4. Can anyone tell me how can i call t_relay() from another module. Can anyone help me? any suggestion? Thanks for your help. Regards -- Sabino Frisardi

3 2

[Serusers] how can i get the IP?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip ?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 03 Sep '08

03 Sep '08

Thanks David for your answer, but can you explain how radius can do that since sending BYE messages is more at the signaling level. I dont quite understand how radius can even communicate at the SIP signaling level. From: David Villasmil Sent: Sunday, August 31, 2008 4:33 PM To: Matteo D'Amato Subject: Re: [Kamailio-Users] CDRTool and Prepaid You could use radius. On Sun, Aug 31, 2008 at 10:09 PM, Matteo D'Amato > wrote: Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

9 19

[Serusers] SER 0.9.7-pre3 and timers
by Steven C. Blair 02 Sep '08

02 Sep '08

I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. Thanks, Steve

1 0

[Serusers] Error opening ser's fifo /tmp/ser_fifo
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 6.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

2 1

[Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu 8.04LTs
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 8.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744 -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

1 0

[Serusers] Local number portability
by inge 02 Sep '08

02 Sep '08

Hi all, Is someone know if it's possible to use SER for prefix calls for LNP ? Then the call will be sent to AS5350 and the prefix must define the carrier for out ported number. I search information on this topic but I find nothing about SIP and LNP. Thanks for our informations. Regards

1 0

[Kamailio-Users] route number
by BERGANZ François 02 Sep '08

02 Sep '08

Hello, I need to have a route number like route(1000) But after some tests, I cant have more than route(39) How can I have more? Thank you

2 1

[Kamailio-Users] How to config OpenSER as IMS server
by Steven Wu 01 Sep '08

01 Sep '08

Hi everybody, Currently we are working on an IMS project. This project need a IMS server. Finally we choiced OpenSER as the IMS server. But have no experience how to use OpenSER as a IMS server. Could someone tell me how to config it? Best regards, Steven Wu

3 2

[Kamailio-Users] Mediaproxy 2.0 - Dispatcher vs Relay
by Nuno Marques 01 Sep '08

01 Sep '08

Hi, I'm trying to understand better how mediaproxy works. Just read the mediaproxy-ng.org page... i understood the necessity of using mediaproxy for NAT traversal, but i have one doubt... - What's the difference between the dispatcher and the relay? Can someone explain me the concept or point me to some literature so i can clear my mind :) Thanks in advance, Nuno

3 2

[Kamailio-Users] kamailio 1.5 and beyond
by Henning Westerholt 01 Sep '08

01 Sep '08

Hi all, i want to share some thoughts where i think the priorities for the next major release(s) of Kamailio should be. This is the continuation of the work i (and many others) have done for the previous releases, but i thought it makes sense to summarize it a bit on a higher (abstraction) level. 1. Refactoring and cleanup We still have quite a lot of old code in our repository, which is no longer working or obsolete. Some implementations uses way to much gotos, or macros that are several pages long. This makes important parts of the core and modules harder to understand and maintain then necessary. Especially in the area of utility functions there is still to much duplicated code. proposed actions: - identification and removal of no longer working code in the core and modules - identification and unification of redundant (duplicated) code in core and modules to one implementation - identification and removal of obselete code parts in core and modules. - refactoring of code to use less gotos and macros 2. User and developer documentation With regards to user documentation quite a lot of work has been done in the past releases. But with the developer documentation, especially in certain modules, there is still plenty of content missing. As the involvement of individuals and companies can change, documentation is necessary to ensure a continued maintenance of the code. proposed actions: - conversion of already existing documentation to doxygen format - add a short doxygen module definition to every module, explaining the purpose - add a short doxygen file definition to every file, explaining the purpose - extend doxygen docs in important areas like the core and modules like TM - autogeneration of database related module documentation from the XML scheme definitions 3. Consolidation in core and modules At the moment we've 85 modules in our repository. Some modules implement only one or two functions, which makes no real sense with regards to the maintainance overhead that comes with it. 24 of the modules are using database access and 7 implement database interfaces. There is no clear boundary in some areas of the core and modules, this hinders the further development of the server. proposed actions: - integrate several modules that share responsibilities and not implement that much functionality on their own into one - autogeneration of module interface and database infrastructure code from the XML database scheme definitions - identify obselete module functions that can be done easily with pseudo-variables now. Remove these functions. - move certain core parts that do not need to be there into modules, making the core smaller and more flexible - identify core APIs which are used from modules, evaluate eventual generalisations to make the modules less dependent from the actual core implementation 4. Quality assurance The complexity of several modules, and the special dependencies that some of them carry increase the work to test the server and reproduce bugs. Without proper test coverage maintenance work is slowed down and flexibility is lost. All "easy" bugs that get caught automatically will not reach the user, increasing the quality. proposed actions: - extend available test suite to cover more of core and modules - continue maintenance of available infrastructure for build tests (build bot, debian package autobuilder), evaluate extensions to increase coverage All this work will be done as in the past in an open and accountable way on the developer list. For bigger changes a request for comment will be posted in advance on this list. If existing functionality is deprecated, then this RFC will be also posted to the user list. If some functionality is moved or changed than this will be documented in the wiki (porting docs) as usual. I understand some changes will be probably add work on your side, in porting, configuration adjustments and the like. But i think this is necessary to ensure the further development of the Kamailio project, and to be able to deal with new challenges that will come up in the future. Any input and help with this regards to this individual tasks is of course much appreciated. In the end the success of the project depends in the future (like it did in the past) on your support. Without your involvement in contributions, discussions and bug reports i would be only to a much lesser extend involved in this project. Thank you, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

2 2

[Kamailio-Users] About Changing 401 Message
by Ali Jawad 31 Aug '08

31 Aug '08

Hi All When I do logon with a wrong username/pwd I get the following error 401 UNAUTHORIZED. If I do want to change that error "for whatever stupid reason :-)" I know I could change that message using the header files and recompile. However I only know where to change the UNAUTHORIZED keyword, what about the 401 number...apart from that could I change that message using ss_reply ? Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

4 6

Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 31 Aug '08

31 Aug '08

Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo

1 0

Re: [Kamailio-Users] [Kamailio-Business] project foundation
by Daniel-Constantin Mierla 31 Aug '08

31 Aug '08

Hello, On 08/13/08 08:37, Max van Biezen wrote: > Hi, > > >> a quick operational expenses estimation for the first year brought up >> the figure of ~3000Euro, that includes the expenses for creating the >> foundation, two hosted servers (one main, one backup), domain names and >> trademark registrations. The costs regarding TM registration expenses >> (US&EU) are not yet very clear to me, maybe someone can shed some light >> if he/she did it recently. >> > > I can only give an indication on EU trademark. That is 850,- one time > fee. This fee is due regardless the fact some company will complain or > not. As happened to me. > > The process can be done by yourself no lawyers needed. However there are > lawyers outhere that simply complain to everything in the hope some > company which they claim to represent willl pay for it. Anyway worst > case scenario would then be to drop the registration whihc means the 850 > is gone > > Bottom line just try it yourself. It is a long process before > registration is final (more than 6 months) > thank you for the details. Cheers, Daniel > Max van Biezen > > > -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] issue when adding subscriber to mysql
by Alex R.S.M 30 Aug '08

30 Aug '08

Hi, Ijust added mysql module to my openser 1.2 and tried to test it by adding a couple of SIP users to the database. I can verified there is only one default sub in the database: mysql> select count(*) from subscriber; +----------+ | count(*) | +----------+ | 1 | +----------+ 1 row in set (0.00 sec) mysql> show tables; I can add the first subscriber: # openserctl add 100 100 100@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 -e mysql_query -e insert -e 0 -e new user '100' added Howver, When I try to add the second subscriber I am getting the following error: # openserctl add 101 101 101@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 -e mysql_query -e insert -e 0 -e \E[37;31mERROR: introducing the new user '101' to the database failed root@bling1:/etc/openser# What might be the problem? Thanks, Alex _________________________________________________________________ See what people are saying about Windows Live. Check out featured posts. http://www.windowslive.com/connect?ocid=TXT_TAGLM_WL_connect2_082008

2 1

[Serusers] SER Version
by Arun Kumar 30 Aug '08

30 Aug '08

Hi All, I'm planning to roll out a new production SER for my company needs suggestion on SER version. At present I run 0.9.6 and having some issue like waiting transaction, extra connection to mysql, so which version should I roll out. thanks arun

4 9

[Kamailio-Users] openser children parameter
by toly 29 Aug '08

29 Aug '08

Greetings, What would be value of children parameter for few thousands user? Thanks, Toly -- View this message in context: http://www.nabble.com/openser-children-parameter-tp19192868p19192868.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 4

[Serusers] AVP and MySQL question
by Steven C. Blair 29 Aug '08

29 Aug '08

I have a SER 0.9.7-pre3 system. I use avopops to load flags from mysql to implement call andling features. I'd like to read a list of values for a single attribute from the database (and I can today) however the order in which the results are return is not consistent. I read that MySQL4 has an issue where results of a query are returned in a nondeterministic fashion. I think I can get my desired result by issuing my own query but I do not see an avpops function which allows for a user specified query. Does such a function exist? If so what is it's name? Thanks

2 1

[Kamailio-Users] Question regarding outbound configuration.
by Siddhardha Garige 29 Aug '08

29 Aug '08

Hello all, I am working on an outbound proxy configuration and here is my setup. UA ------> Outbound ------->firewall(NAT) -------->Proxy. Both UA and outbound are behind the NAT. I did some RFC study on outbound configuration for routing calls both inbound and outbound through outbound proxy. According to RFC 3327 Extension Header field for Registering Non-Adjacent contacts, PATH variable has to be introduced into REGISTRATIONS. Here are few questions and any help with answers is greatly appreciated. 1. Default openser.cfg will route request to destination if (URI != myself) and appends p-hint:outbound. What exactly is the significance of P-hint:outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 says ob parameter has to be added to PATH filed or Contact Field to retain path. P-hint got to do any thing regarding this? 2. I enabled path.so module. But openser does not at PATH filed to outgoing requests. Can someone direct me to a config example? Thanks in advance, Sid

1 1

[Kamailio-Users] event notifications and user aliases
by Alexander 'Leo' Bergolth 28 Aug '08

28 Aug '08

Hi! I'd like to implement P2P presence and extension monitoring for my SIP network. Our users register by using a numeric extension (e.g. sip:226050@wu-wien.ac.at), but most of the time, they are contacted using a user alias that looks like their email-address (e.g. sip:bergolth@wu-wien.ac.at). So the addresses stored in their buddy-lists won't be the "real"-SIP-URI of the users. Will there be a problem if I translate those aliases on subscription, so that a user may also subscribe for the presence of a user alias? If I rewrite SUBSCRIBE requests to use the "real" SIP-URI on subscription, the following NOTIFY messages will contain the wrong URI in the From-header and in the pidf body. But maybe this won't be a problem, since the watcher can assign it to the correct username because the NOTIFY belongs to the same dialog as the SUBSCRIBE? Will the user agents be able to correctly assign those notifications to their contacts? Any experiences with this? Cheers, --leo -- e-mail ::: Leo.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

4 4

[Serusers] Call forwarding: Diversion header
by caio 28 Aug '08

28 Aug '08

Hi, Does anyone know how to implement draft-levy diversion header on for example bilnd call forwarding? I have this scenario: xlite ------ SER-0.9.7 ------ PSTN-GW ------ PSTN Case: Call from xlite-579950 to 579917 (both registered at SER), 579917 has fwdblind enabled to pstn-501866. where SER modifies INVITE when detect that user 579917 has fwdblind feature enabled. The call must be forwarded to the pstn number. but the PSTN-GW returns to SER a "302: Moved Temporarily" with a confused diversion header.. Here a few sip messages: = SER -----> PSTN-GW U +0.000824 200.2.127.187:5060 -> 200.2.127.153:5060 INVITE sip:501866@200.2.127.153:5060 SIP/2.0. Record-Route: <sip:200.2.127.187;ftag=2d0ad32d;lr=on>. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. Max-Forwards: 16. Contact: <sip:579950@192.168.132.22:35318>. To: "579917"<sip:579917@wsa.lab>. From: "579950"<sip:579950@wsa.lab>;tag=2d0ad32d. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 385. = PSTN-GW ----> SER U +0.051198 200.2.127.153:5060 -> 200.2.127.187:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. To: "579917" <sip:579917@wsa.lab>;tag=121881197764d164f2-9431-47bb-be95-8af6684ebc04. From: "579950" <sip:579950@wsa.lab>;tag=2d0ad32d. CSeq: 2 INVITE. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. Contact: <sip:579917@200.2.127.187:5060>. Diversion: <sip:501866@200.2.127.153:5060>;reason=unknown. Server: Epygi Quadro SIP User Agent/v4.1.23 (SOHOGW). Content-Length: 0. What is wrong? How the flow must be to satisfies diversion of levy draft? Any comments are welcome...i'm confused if SER must modify INVITE or "To" header or if a diversion must be set (add_diversion of diversion.so module) when SER send the first invite to PSTN-GW... Regards, -- caio

1 0

[Kamailio-Users] Why dialplan and drouting modules were removed from openser-1.3.3?
by Peter Lemenkov 28 Aug '08

28 Aug '08

Hello All! Subj. -- With best regards!

3 4

[Kamailio-Users] How to test RLS and trace messages?
by mahesh.peddi＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi, I am working on RLS Application server and integrated with SER IMS, i made changes in the configuration file with the help of Readme file. Its working fine. But how to confirm and test RLS is working. And how to trace the messages? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] Using pua_bla for BLF
by catalina oancea 28 Aug '08

28 Aug '08

Hello I am trying to make some phones work with BLF using Kamailio and I found an older messge ( http://lists.kamailio.net/pipermail/users/2007-October/013700.html) saying that I could do this using BLA. I have a Thomson ST2030 as the message says, but it doesn't work with presence and pua_bla. I get the error: ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: event= dialog I undestood from that message that the event is supposed to be 'dialog', so what's the problem then? Thanks a lot Catalina

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[Kamailio-Users] Fw: regarding 3rd party register request.
by srinivasa.reddy＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi all, This is Srinivas, i am working on OpenIMScore and Openser( as Application Server) . from few days i stuck at third party regaester request(that is interface between SCSCF and Presence ). You said that, the registration-event package is not enabled at the IMS Core side that mean in the scscf.cfg is it? If it is could you please provide where and what i have to change and i am attaching the scscf.cfg . And regarding 2nd point Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package That means openser does not support reg package or ? WITH BEST REGARDS, SRINIVAS. -------------------------------------------------------------------------------------------------------------------------------- Hi, >From your mail below, I could infer only that, the registration-event package is not enabled at he IMS Core side. Make sure, that you have done the necessary settings for that. Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package. Thanks, Noble Antony T. On Wed, Aug 27, 2008 at 3:54 PM, <mahesh.peddi(a)info-spectrum.com> wrote: Hi, I have one issue regarding openser could you kindly reply if you know . I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am connected it with IMS core .And for testing i was used UCTIMS client during the register with uctims client actually i need to get 4 messages. 1.Third party register request is going to presence server from scscf. 2.Notify messages coming from presence server to uctclient. 3.subscribe message must take route through the presence server. 4.publish message must take route through the presence server. But at present i am getting 3 and 4 messges corectly i am not getting 1 and 2. So please tell me what are the modifications or changes i need to do in any xml files or cfg files. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764 Regards, Srinivasa Reddy Kadiri Software Developer Infospectrum India Pvt. Ltd. Cell: +91 9921710062 Email: srinivasa.reddy(a)info-spectrum.com IP-Phone Ext. - 759

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Re: [Kamailio-Users] How to test and trace the RLS messages?
by Daniel-Constantin Mierla 28 Aug '08

28 Aug '08

Hello, please keep cc-ing to mailing lists, there are people that can answer you quicker and better. As a matter of fact, although I have studied a bit IMS docs, that happened long time ago and don't know how those messages should be routed in IMS and where the mistake can be in your routing logic. Cheers, Daniel On 08/27/08 13:24, mahesh.peddi(a)info-spectrum.com wrote: > Hi, > > I have one issue regarding openser could you kindly reply if you know . > > I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am > connected it with IMS core .And for testing i was used UCTIMS client > during the register with uctims client > actually i need to get 4 messages. > 1.Third party register request is going to presence server from scscf. > 2.Notify messages coming from presence server to uctclient. > 3.subscribe message must take route through the presence server. > 4.publish message must take route through the presence server. > But at present i am getting 3 and 4 messges corectly i am not getting > 1 and 2. > So please tell me what are the modifications or changes i need to do > in any > xml files or cfg files. > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: "userlist" <users(a)lists.kamailio.org> > Sent: Wednesday, August 20, 2008 10:44 PM > Subject: Re: [Kamailio-Users] How to test and trace the RLS messages? > > >> Hello, >> >> On 08/20/08 09:44, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> I am working on openser 1.3.2 version with RLS as >>> "integrated_xcap_server". Its running without erros. >>> But my query is how to test, And please send me call flow between >>> IMS components and RLS server. >> perhaps you can find some hints in IMS specifications. You need in >> first place a sip phone with rls support, afaik, there are not that >> many. >> >> Cheers, >> Daniel >> >>> Regards, >>> Mahesh Peddi >>> Infospectrum India Pvt. Ltd. >>> Cell: +91 9765775176 >>> IP-Phone Ext. - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] openser v1.3.3. released
by Daniel-Constantin Mierla 27 Aug '08

27 Aug '08

Hi everybody, OPENSER v1.3.3 is out - it is a minor release of the branch 1.3, that includes the fixes since v1.3.2 committed so far- configuration file and database compatibility is preserved. OPENSER 1.3.3 is based on the latest version of branch 1.3, therefore those running 1.3.2 or earlier version in this branch are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/openser/1.3.3/src/ Detailed changelog: http://www.kamailio.org/pub/openser/1.3.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 openser Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.3.3/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] Radius Auth : 407 proxy auth and Get-User-Attr - crash
by Samuel Muller 27 Aug '08

27 Aug '08

Hello all, i'm using SER 2.0 with Radius accounting and authentication - based on x64 ubuntu machine. i've a problem with the Radius auth, and only for the proxy auth mechanism (method INVITE) : depending of the topology of the UAC, it crashes the SER or a there's a 407 proxy auh required. when the UAC is behind a simple NAT, i get the 407, with "preauth: credentials with realm 'sip.720.fr' not found" - i tried many configs, no way it doesn't work. but the REGISTER is ok - it's the same Radius request ? when the UAC is behind NAT but in a VPN (our customers), proxy auth is successfull even if the preauth don't find the credentials with the realm (why? it does not differ from another uac without VPN ???), but when SER requests the Radius to get the user attributes, it crashes ! in each case, there's no pbm with REGISTER requests. Radius user sample : 1010(a)sip.720.fr User-Password = "pwd" User-Name = "1010", SER-UID = "1010(a)sip.720.fr", SER-DID = "1", SER-Digest-Username = "1010", SER-Digest-Realm = "sip.720.fr", SER-Uri-User = "1010(a)sip.720.fr" Does anyone have any idea, or met already this problem ? What I have to verify ? Thanks for help, Samuel MULLER

1 0

[Kamailio-Users] multiple extensions/domains
by voipmo 27 Aug '08

27 Aug '08

I have been running asterisk servers for a few years now to host multiple organizations. I have been working on setting up OpenSER so that we can have failover and load balancing. My question is how do I keep my different organizations separate? I currently have the following registrations: 1001(a)test.example.com 1002(a)test.example.com 1001(a)another.example.com The problem is that if 1002(a)test.example.com dials 1001 it rings both domains 1001 phones. Am I going about this the right way? Is there a better method? How might I prevent calls between domains? -- View this message in context: http://www.nabble.com/multiple-extensions-domains-tp19173400p19173400.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Serusers] Help with Ser and LCR
by Jai Rangi 26 Aug '08

26 Aug '08

All, I need little help with LCR module in ser, Say these are the entries on my LCR tables, Prefix, | From_URI | Grp_id | Priority "13612499080" ;"%"; "46"; "1" "1" ;"%"; "1"; "1" "1" ;"%" "2" "2" Now any Say if someone calls 19495556666, Then ser sends the call to Server in Grp_id 1, if the grp_id=1 server does not respond, then SER sends the call to Server with grp_Id 2. Perfect, that exactly what I need. Now if some calls 13612499080", then SER will send the call to server with grp_id 46. If the call failes it should not send the call to server grp_id 1 or two. Right? In my case it is sending the call to server grp_id 1 after the call fails on server grp_id 46. How can I stop that. I want the calls for 13612499080 forwarded to only server grp_id 46 and is that fails then just terminate the call. I will appreciate if someone can provide any help or hint. Thank you, -JP

1 1

[Kamailio-Users] AVP to store user contact information
by Asim Riaz 26 Aug '08

26 Aug '08

Hi All, I am wondering if there is a way to store the user contact info into avp, relay it to existing Regitrar and store the original contact info into location through a function(which i dont know) once i received successful reply from my sip registrar. Thanks in advance Regards Asim Riaz

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[Kamailio-Users] FW: | SMS FALAS | <--- did anyone else on this list get this message?
by Vadim Berezniker 26 Aug '08

26 Aug '08

Sorry this is offtopic but.. Did anyone else get this email? I'm just wondering if they scraped emails from this mailing list. -----Original Message----- From: KLIENT I NDERUAR [mailto:info@hi5.com] Sent: Tuesday, August 26, 2008 11:38 AM To: Vadim Berezniker Subject: | SMS FALAS | Klient i/e Nderuar Nese Deshironi te Ndegjoni Muzika ma te rejat ather vizitone Kosova-SMS dhe do te bindeni me muziken qe ofrojm ne. Gjithashtu per Vizitoret e Kosova-SMS ka Startu me ofert te RE. SMS FALAS MUNDSUAR NGA Kosova-SMS 60 SMS NE DIT NGA NR I YT KLIKO PER ME SHUM. --> http://www.kosova-sms.tk

1 0

[Serusers] Notice call ending
by inge 26 Aug '08

26 Aug '08

Hi all, I look for a workarround to notice the caller that the call will be hang up, by generating a beep. Is someone can help me because I don't find anything about this subject ? Thanks. Regards, Adrien.

1 0

[Kamailio-Users] Having Problem routing an ACK
by Stagg Shelton 26 Aug '08

26 Aug '08

Hello, I am having an issue routing an ACK back from a new upstream carrier. The basic situation is my openser 1.3.2 receives the INVITE. I do an alias_db_lookup, set some accounting flags, and send the call to t_relay The call makes it to the PBX, as expected, and the PBX sends a 100 trying, 180 ringing. The 180 ringing is sent to the carrier. When the incoming call is answered the PBX sends a 200OK to openser, and that 200OK is sent to the carrier. When the carrier sends the ACK back it makes it to OpenSER, but is not sent to the PBX. The PBX will send several more 200 OK attempts, but each ACK from the carrier stops at openser. From what I can tell from the wireshark packet captures, the Request- Line header in the ACK from the carrier is sip:TELNUMBER@MY_OPENSER_IPADDR I have compared these packets to similar traces from other carriers and the Request-Line header from the other carriers is sip:TELNUMBER@MY_PBX_IPADDR This new carrier is telling me that they are not doing anything wrong with the ACK. RFC3261 seems specific about the construction of the ACK, but at the same time i am getting lost going to all of the other sections trying to identify details. Basically I just need to get the ACK to the PBX. What other information can I provide with this problem. Thanks Stagg Shelton

4 20

[Serusers] ser 0.10 doucmentation
by Steven C. Blair 26 Aug '08

26 Aug '08

I was wondering if anyone has documentation on the new syntax used in SER 0.10.x. I see the following in the ser-advanced.cfg example file but I do not see these parameters documented anywhere. Thanks,Steve if ($t.fr_inv_timer) { if ($t.fr_timer) { t_set_fr("$t.fr_inv_timer", "$t.fr_timer"); } else { t_set_fr("$t.fr_inv_timer"); } } Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

6 32

[Kamailio-Users] CDRTool not applying rates
by Brian Del Shasta 25 Aug '08

25 Aug '08

I believe my rating tables are configured correctly now thanks to David Villasmil who sent me examples from his working setup. However, I am still unable to see any rates. The rating engine seems to be started and I have no problem identifying and matching calls based on specific criteria. But I still see no rates. I'm wondering if anyone knows the best place to troubleshoot the rating engine would be or if anyone has any other ideas? Thank you in advance, and thanks again David for your help!

3 7

Re: [Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

in this guide, he doesn't have the module presence_xml, but he has the parameter modparam (" presence ", " force_active ", 1) http://www.kamailio.net/dokuwiki/doku.php/presence:configuration-file it is strange, I believe to be seen in the wiki a same example for openser 1.3.2 greetingss rickygm

2 1

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