sr-users September 2004

sr-users@lists.kamailio.org

190 participants
382 discussions

[Serusers] avp.so test results
by Java Rockx 30 Sep '04

30 Sep '04

Hello All. I'm using ser-0.8.99-dev7 and testing the new AVP module. I check this code out of CVS this morning and it compiled just fine. However, I'm getting a loadmodule error. [root@sip01 ser]# cat /var/log/messages Sep 30 10:33:50 sip01 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/avp.so>: /usr/local/lib/ser/modules/avp.so: undefined symbol: db_free_query Sep 30 10:33:50 sip01 ser: parse error (99,13-47): failed to load module Sep 30 10:33:50 sip01 ser: ERROR: bad config file (1 errors) Sep 30 10:33:50 sip01 ser: ser startup failed Cheers, Paul __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

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[Serusers] ser.cfg tuning question
by Java Rockx 30 Sep '04

30 Sep '04

Hello. I'm using MySQL with ser. I need to check the ACL several places in my ser.cfg script. Is there any difference in calling is_user_in() multiple times versus calling it once and setting a flag. Then referencing the flag elsewhere rather than calling is_user_in() again? I guess I just don't know if calling is_user_in() hits the database every time or if it caches results. Example: if (is_user_in("Request-URI", "voicemail")) { setflag(31); }; some where else in ser.cfg: if (isflagset(31) { do something }; Regards, Paul _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com

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[Serusers] User Agent Behind Nat is not sending ACK! Help, Please!
by armando 30 Sep '04

30 Sep '04

Hi! I have a very weird trouble here... I've been watching using a sniffer the messages exchange between 2 UA and SER .... and I can see very clear that the Caller UA never reply with an ACK message after it has received the 200 OK message from the Callee UA. It just open the rtp port and start sending out rtp streams... I have tried this either using ATA and Sipuras, but always the same result... because of this the caller never received audio streams from the callee due this last one is waiting for the ACK message after it sent out the 200 OK message... I have no clue why they are no sending this message to the callee.. I need help please!!! Any help or clue or hint...or whatever will be really appreciate!!! Thanks in advance! Armando Marrero Cti, Miami, FL Here is a Copy of a ngrep session. This ngrep session is in the attached file too. Bob using Real IP , Armando Private IP behind NAT.... rtpproxy is enforced. Call 1 = From Bob -> Armando = Success Call 2 = From Armando -> Bob = Failed because Armando UA never reply with ACK after 200 Ok Message ############ Call 1 caller: Bob , Callee : Armando.... Successfull Call Setup U Armando -> SER SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.90.116;branch=z9hG4bK1ba9.f6fe0847.0 .Via: SIP/2.0/UDP 66.231.238.215:5060..Record- Route: <sip:3056034425@10.1.90.116;ftag=1564176202;lr=on>..From: <sip:1234@10.1.90.116;user=phone>;tag=1564176202..To: <s ip:3056034425@10.1.90.116;user=phone>;tag=642782702..Call-ID: 2352639743@66.231.238.215..CSeq: 1 INVITE..Contact: <sip:30 56034425@192.168.0.103:5061;user=phone;transport=udp>..Server: Cisco ATA 186 v2.16 ata18x (030509a)..Content-Length: 203 ..Content-Type: application/sdp....v=0..o=3056034425 62077 62077 IN IP4 192.168.0.103..s=ATA186 Call..c=IN IP4 192.168.0. 103..t=0 0..m=audio 16384 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. # U SER -> Bob SIP/2.0 200 OK..Via: SIP/2.0/UDP 66.231.238.215:5060..Record-Route: <sip:3056034425@10.1.90.116;ftag=1564176202;lr=on>..F rom: <sip:1234@10.1.90.116;user=phone>;tag=1564176202..To: <sip:3056034425@10.1.90.116;user=phone>;tag=642782702..Call-ID : 2352639743@66.231.238.215..CSeq: 1 INVITE..Contact: <sip:3056034425@66 176.127.120:5061;user=phone;transport=udp>..Serv er: Cisco ATA 186 v2.16 ata18x (030509a)..Content-Length: 242. Content-Type: application/sdp....v=0..o=3056034425 62077 62077 IN IP4 192.168.0.103..s=ATA186 Call..c=IN IP4 66.231.238.115..t=0 0. m=audio 35118 RTP/AVP 0 101..a=rtpmap:0 PCMU/8 000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15. a=direction:active..a=nortpproxy:yes.. # U Bob -> SER ACK sip:3056034425@10.1.90.116 SIP/2.0..Route: <sip:3056034425@66.176.127 120:5061;user=phone;transport=udp>..Via: SIP/2. 0/UDP 66.231.238.215:5060..From: <sip:1234@10.1.90.116;user=phone> tag=1564176202..To: <sip:3056034425@10.1.90.116;user=p hone>;tag=642782702..Call-ID: 2352639743@66.231.238.215..CSeq: 1 ACK. User-Agent: Cisco ATA 186 v2.15 ata18x (030313a).. Content-Length: 0.... # U SER -> Armando ACK sip:3056034425@66.176.127.120:5061;user=phone;transport=udp SIP/2.0. Max-Forwards: 10..Record-Route: <sip:3056034425@ 10.1.90.116;ftag=1564176202;lr=on>..Via: SIP/2.0/UDP 10.1.90.116;branch=0. Via: SIP/2.0/UDP 66.231.238.215:5060..From: <s ip:1234@10.1.90.116;user=phone>;tag=1564176202..To: <sip:3056034425@10.1 90.116;user=phone>;tag=642782702..Call-ID: 23526 39743@66.231.238.215..CSeq: 1 ACK..User-Agent: Cisco ATA 186 v2.15 ata18x (030313a)..Content-Length: 0.... ################## Call 2 caller: Armando , Callee : Bob.... Failed Call Setup .... U Bob -> SER SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.90.116;branch=z9hG4bK4544.49cf0507.0 .Via: SIP/2.0/UDP 192.168.0.103:5061;received= 66.176.127.120..Record-Route: <sip:1234@10.1.90.116;ftag=298050148;lr=on>. From: <sip:3056034425@lserver.telecode.com;use r=phone>;tag=298050148..To: <sip:1234@lserver.telecode.com;user=phone> tag=1758946682..Call-ID: 2670895917(a)192.168.0.103. .CSeq: 1 INVITE..Contact: <sip:1234@66.231.238.215:5060;user=phone transport=udp>..Server: Cisco ATA 186 v2.15 ata18x (0 30313a)..Content-Length: 197..Content-Type: application/sdp....v=0..o=1234 1415 1415 IN IP4 66.231.238.215..s=ATA186 Call ..c=IN IP4 66.231.238.215..t=0 0..m=audio 16384 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000. .a=fmtp:101 0-15.. # U SER -> Armando SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.103:5061;received=66.176.127 120..Record-Route: <sip:1234@10.1.90.116;ftag=298 050148;lr=on>..From: <sip:3056034425@lserver.telecode.com;user=phone> tag=298050148..To: <sip:1234@lserver.telecode.com;u ser=phone>;tag=1758946682..Call-ID: 2670895917@192.168.0.103..CSeq: 1 INVITE..Contact: <sip:1234@66.231.238.215:5060;user =phone;transport=udp>..Server: Cisco ATA 186 v2.15 ata18x (030313a). Content-Length: 235..Content-Type: application/sdp. ...v=0..o=1234 1415 1415 IN IP4 66.231.238.215..s=ATA186 Call..c=IN IP4 66 231.238.115..t=0 0..m=audio 35122 RTP/AVP 0 10 1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=direction:active..a=nortpproxy:yes.. # ------ no ACK send back....in this moment Armando UA just open rtp and start sending voice streams ------ Bob User Agent resend this several times , but never get ACK message from Armando UA U Bob -> SER SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.90.116;branch=z9hG4bK4544.49cf0507.0 .Via: SIP/2.0/UDP 192.168.0.103:5061;received= 66.176.127.120..Record-Route: <sip:1234@10.1.90.116;ftag=298050148;lr=on>. From: <sip:3056034425@lserver.telecode.com;use r=phone>;tag=298050148..To: <sip:1234@lserver.telecode.com;user=phone> tag=1758946682..Call-ID: 2670895917(a)192.168.0.103. .CSeq: 1 INVITE..Contact: <sip:1234@66.231.238.215:5060;user=phone transport=udp>..Server: Cisco ATA 186 v2.15 ata18x (0 30313a)..Content-Length: 197..Content-Type: application/sdp....v=0..o=1234 1415 1415 IN IP4 66.231.238.215..s=ATA186 Call ..c=IN IP4 66.231.238.215..t=0 0..m=audio 16384 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000. .a=fmtp:101 0-15.. # U SER -> Armando SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.103:5061;received=66.176.127 120..Record-Route: <sip:1234@10.1.90.116;ftag=298 050148;lr=on>..From: <sip:3056034425@lserver.telecode.com;user=phone> tag=298050148..To: <sip:1234@lserver.telecode.com;u ser=phone>;tag=1758946682..Call-ID: 2670895917@192.168.0.103..CSeq: 1 INVITE..Contact: <sip:1234@66.231.238.215:5060;user =phone;transport=udp>..Server: Cisco ATA 186 v2.15 ata18x (030313a). Content-Length: 235..Content-Type: application/sdp. ...v=0..o=1234 1415 1415 IN IP4 66.231.238.215..s=ATA186 Call..c=IN IP4 66 231.238.115..t=0 0..m=audio 35122 RTP/AVP 0 10 1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=direction:active..a=nortpproxy:yes.. # U Bob -> SER SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.90.116;branch=z9hG4bK4544.49cf0507.0 .Via: SIP/2.0/UDP 192.168.0.103:5061;received= 138,3 56%

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[Serusers] SER, NAT, RTPPROXY
by Grzegorz Kowalczyk 30 Sep '04

30 Sep '04

Hi, Is there anybody here who has a such ser configuration, which can serve properly all events of RTP streams? Which can resolve type of NATs behind which both UAs are and if it is neccessary can redirect RTP data across the rtpproxy or if it is not neccessary (e.g. FULL CONE NAT or public IP) the rtpproxy will not be used? G.

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Re: [Serusers] SER + rtpproxy
by Andrei Pelinescu-Onciul 30 Sep '04

30 Sep '04

On Sep 29, 2004 at 13:33, Soren (Home) <soren(a)tanesha.net> wrote: > Hi Andrei, > > The problem to me is, that almost never it returns true, since most > clients have buildin STUN today. > > However, X-lite, Grandstream and some others I've tried all send the > public IP address also for Symmetric NATs, hence Client<->Client will not > work without rtpproxy. I thought X-Lite was one of the few which got it right. > > Any suggestions on how to solve that with nat_uac_test ? Try the attached patch. It adds a new nat_uac_test flag (16) that will test for differences between the source port of the message and the port in via. Usage: nat_uac_test("16") or to include all the common tests: nat_uac_test("19") ( private ip in contact | src ip != ip in via | src port != port in via) (like all the other via tests, it makes sense only for requests) Andrei

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[Serusers] SER/asterisk integration?
by Roy Sigurd Karlsbakk 30 Sep '04

30 Sep '04

hi after trying to use asterisk as a SIP frontend, I've given up, and trying to make SER do the job. It should be a quite simple setup, but I have a few questions - How can I specify a database host and user with serctl? - Are there any scripting support as in Asterisk AGI? thanks roy

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[Serusers] acc module bug ??
by Java Rockx 30 Sep '04

30 Sep '04

Hello All. I've got a question about the acc module accounting for missed calls. Right now missed call records are recorded the ser/missed_calls table in my MySQL database. serweb does not show missed calls. I turned on SQL tracing and identified why serweb is not showing the records from the missed_calls table. The SQL being executed is as follows: (SELECT t1.from_uri, t1.sip_from, t1.time, t1.sip_status FROM missed_calls t1 WHERE t1.username='1002' and t1.domain='mycompany.com' ) UNION ( SELECT t1.from_uri, t1.sip_from, t1.time, t1.sip_status FROM missed_calls t1, aliases t2 WHERE 'sip:1002@mycompany.com'=t2.contact AND t2.username=t1.username AND t2.domain=t1.domain) The problem that the missed_calls.username and missed_calls.domain columns are NULL which causes the query to return zero rows. Is this a bug in the acc module, or have I forgotten to do something in ser.cfg? My acc module params are as follows: modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_fmt", "fisum") modparam("acc", "report_ack", 1) modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 2) modparam("acc", "db_url", "mysql://ser:pwd@sipdb01.mycompany.com/ser") I'm using ser 0.8.99-dev6 Regards, Paul _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com

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RE: RE: [Serusers] mediaproxy problem
by Zeus Ng 30 Sep '04

30 Sep '04

Kun, Without ser.cfg and ngrep dump from the ser server, the only thing I can say is your nat handling logic is not properly setup. SER try to forward your INVITE request to callee on the port that was blocked by the NAT device. Zeus > -----Original Message----- > From: ?￡à¤ [mailto:haoniukun@sohu.com] > Sent: Thursday, 30 September 2004 6:38 PM > To: Zeus Ng > Subject: Re:RE: [Serusers] mediaproxy problem > > > > Dear Zeus, > > Thanks for your reply, > Yes, I make myself understand. I'm trying to call myself. > I'm using the mediaproxy. But I don't know exactly why a user > behind a firewall can't contact a user behind another > firewall. I can get the 404 error if the user doesn't exist. > But when I call some user exist like myself, I always get 100 > trying message, after i get 408 error. Would you mind tell me > how this problem would happen? > > Thanks in advance. > > Sincerely, > Kun > > > ----- 原文 ----- > From: Zeus Ng > To: '?￡à¤' > Cc: serusers(a)lists.iptel.org > Subject: RE: [Serusers] mediaproxy problem > Sent: Wed Sep 29 20:38:32 CST 2004 > > > > > Are you saying you can't call yourself? Is it the question > you want to > > ask? > > > > The contact address <sip:8103@ipaddress1:5060> is either > wrong or you > > have made some mistake in obfuscation. It should be > > <sip:8103@ipaddress2:5060>. For former case, correct that address > > first. For the second case, send in the ser.cfg and dump without > > obfuscation. BTW, your IP obfuscation is somewhat broken as > we can see > > that the client private IP is 192.168.1.13 and the public IP is > > 219.137.9.30. > > > > Zeus > > > > > -----Original Message----- > > > From: serusers-bounces(a)lists.iptel.org > > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ?￡à¤ > > > Sent: Wednesday, 29 September 2004 5:55 PM > > > To: serusers(a)lists.iptel.org > > > Subject: [Serusers] mediaproxy problem > > > > > > > > > > > > Dear sirusers, > > > The following is my log file kept by x-lite. > > > > > > SEND TIME: 15466254 > > > SEND >> ipaddress1:5060 > > > INVITE sip:8103@ipaddress1 SIP/2.0 > > > Via: SIP/2.0/UDP ip > > > address2:5060;rport;branch=z9hG4bKEA15D2B6122C11D9B04300E04CEE03A2 > > > From: 8103 <sip:8103@ip address1>;tag=1235055289 > > > To: <sip:8103@ip address1> > > > Contact: <sip:8103@ipaddress1:5060> > > > Call-ID: EA15D2B5-122C-11D9-B043-00E04CEE03A2(a)192.168.1.13 > > > CSeq: 52555 INVITE > > > Max-Forwards: 70 > > > Content-Type: application/sdp > > > User-Agent: X-Lite release 1103m > > > Content-Length: 137 > > > > > > v=0 > > > o=8103 15466182 15466249 IN IP4 ipaddress2 > > > s=X-Lite > > > c=IN IP4 ipaddress2 > > > t=0 0 > > > m=audio 8000 RTP/AVP 0 > > > a=rtpmap:0 pcmu/8000 > > > > > > RECEIVE TIME: 15466262 > > > RECEIVE << ipaddress1:5060 > > > SIP/2.0 100 trying -- your call is important to us > > > Via: SIP/2.0/UDP > > > ipaddress2:5060;rport=19465;branch=z9hG4bKEA15D2B6122C11D9B043 > > > 00E04CEE03A2 > > > From: 8103 <sip:8103@ipaddress1>;tag=1235055289 > > > To: <sip:8103@ipaddress1> > > > Call-ID: EA15D2B5-122C-11D9-B043-00E04CEE03A2(a)192.168.1.13 > > > CSeq: 52555 INVITE > > > Server: Sip EXpress router (0.8.14 (i386/linux)) > > > Content-Length: 0 > > > Warning: 392 ipaddress1:5060 "Noisy feedback tells: pid=2307 > > > req_src_ip=219.137.9.30 req_src_port=19465 > > > in_uri=sip:8103@ipaddress2 out_uri=sip:8103@ipaddress2:5060 > > > via_cnt==1" > > > > > > I use the URI to call back to myself. But it just try. I > > > wonder if this happens when the NAT was set improperly. If > > > so,how can I make some changes? > > > > > > Thanks in advance. > > > > > > Sincerely, > > > Kun > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

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[Serusers] setflag question
by Java Rockx 30 Sep '04

30 Sep '04

Hi All. I'm trying to get missed call entries to be stored in my MySQL database. Calls are being logged to my ser/acc table, but the ser/missed_calls table is empty. I've read the ser admin guide section for Accounting, but still have a question. If I have the following in my ser.cfg modparam("acc", "db_url", "mysql://ser:mypwd@mycompany.com/ser") modparam("acc", "report_ack", 1) modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_fmt", "fisum") And in my route plan I have this setflag(1); setflag(2); if (!t_relay()) { sl_reply_error(); }; Why do I not have records in ser/missed_calls when I call a phone and hang up before it answers? Are the setflag() calls correct, or does ser do a bitwise AND which means I would be calling setflag(3)? I'm using ser 0.8.99-dev6. Regards, Paul __________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

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[Serusers] Accessing User Preferences
by Java Rockx 30 Sep '04

30 Sep '04

Hello All. How would one determine user preferences set in the serweb user_interface application from ser.cfg? I'm referring to the "User Preferences" tab when you log in as a normal user. Suppose I log in to the serweb admin console and define a user preference of "Block 900 Access" as a boolean 1 or 0. Then a user selects this option in the serweb user console. In my ser.cfg how can I access this "feature" setting and route the call accordingly? In this case something like if (check_user_preference("Block 900 Access") == 1) { do something } else { do something else }; Regards, Paul _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com

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sr-users September 2004