sr-users September 2006

sr-users@lists.kamailio.org

184 participants
437 discussions

[Serusers] SIP client with video???
by Joao Pereira 06 Sep '06

06 Sep '06

Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software..... Does someone knows about a better software? Regards Joao Pereira

2 2

[Serusers] log ( history of action for one user)
by Legos Engineering 06 Sep '06

06 Sep '06

HI. I wonder, if there is some log (like messages or another) which permit to know the history of one user ( his last calls, when he was called ...). I want to know that in order to add, an history of each user in the webcare. Therefore, if someone can say me, if this log exist and where it is. Thank you.

2 1

[Users] How to redirect based on from_uri and uri
by Darren DeLitizia 06 Sep '06

06 Sep '06

Here is what I want to do. Domestic routing defining interlata and intralata based off of ani and dnis (from uri and uri). Now let's not dive too much into the whole domestic routing thing or we'll be here for the next century. Bottom line is, this what I want to do with openser. 1. Look at ani (from_uri) (NPA - first 3 digits) 2. Look at dnis (uri) - (NPA - first 3 digits) 3. If the NPA (first 3 digits) in ani is in same as the DNIS NPA then redirect 555dnis@ip_address 4. If ANI NPA is in same as DNIS NPA then redirect 999dnbis@ip_address I am redirecting so I can be and stay stateless I am able to redirect but unable to figure out correct syntax for it to look at ani and dnis then redirect appropriately. Thanks all

2 1

[Users] crash
by Helge Waastad 06 Sep '06

06 Sep '06

Hi, in my 1.1.0 version I syddenly got a core dump. Does any one know why? (gdb) bt #0 0x000000304448660b in strftime_l () from /lib64/tls/libc.so.6 #1 0x0000002a95b29de4 in time2mysql (_time=Variable "_time" is not available. ) at utils.c:54 #2 0x0000002a95b2a731 in val2str (_c=Variable "_c" is not available. ) at val.c:147 #3 0x0000002a95b26b13 in print_values (_c=0x643648, _b=0x2a95db5269 "'SIP/2.0 100 Trying\\r\\nVia: SIP/2.0/UDP 217.17.222.4;branch=z9hG4bK1957.c42a1af6.0\\r\\nVia: SIP/2.0/UDP 10.1.93.186:10001;rport=44323;received=217.17.222.6;branch=z9hG4bK2eb74daf34599095\\r\\nFrom: \\\"Har"..., _l=65431, _v=0x7fbfffe990, _n=10) at dbase.c:144 #4 0x0000002a95b27aa4 in db_insert (_h=0x643e08, _k=Variable "_k" is not available. ) at dbase.c:495 #5 0x0000002a97c26bdc in trace_sl_onreply_out (req=Variable "req" is not available. ) at siptrace.c:1078 #6 0x0000002a956b30ad in run_sl_callbacks (req=0x643f28, buffer=Variable "buffer" is not available. ) at sl_cb.c:88 #7 0x0000002a956b3608 in sl_send_reply (msg=0x643f28, code=100, text=0x608ee8 "Trying") at sl_funcs.c:149 #8 0x0000000000409c38 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:701 #9 0x000000000040a87e in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #10 0x000000000040b43d in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #11 0x000000000040b496 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #12 0x000000000040b762 in run_top_route (a=Variable "a" is not available. ) at action.c:89 #13 0x000000000043188c in receive_msg (buf=0x2a97e4d2f8 "\026", len=822, rcv_info=0x7fbffff810) at receive.c:155 #14 0x000000000044fc74 in udp_rcv_loop () at udp_server.c:465 #15 0x000000000041f7f4 in main_loop () at main.c:925 #16 0x0000000000420446 in main (argc=Variable "argc" is not available. ) at main.c:1477 -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 5

[Users] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi users, This is ngrep report regarding my problem; -----------------<when a sip proxy behind NAT and the user registerd to that sip-proxy calls>------------------------- U 82.102.69.105:39871 -> 81.21.33.35:5060 INVITE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 361. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 82.102.69.105.---------------------------------------------> this is the NAT for the sip-proxy , t=0 0. m=audio 5004 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -----------------------------<when my SIP-SERVER standing on public ip recieved the "INV" message from the above"----------------------------------- U 81.21.33.35:5060 -> pstngw:5060 INVITE sip:99106883@pstngw:5060 SIP/2.0. Record-Route: <sip:99106883@81.21.33.35 :5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK0ab.9522bc25.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:82.102.69.105:39871>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 360. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 81.21.33.35.--------------------------------------------->This is my SIP-SERVER standing on public ip (and is sending to pstngw) t=0 0. m=audio 60516 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. ---------------------------------------------<and the pstngw is making call>----------------------------------------------- U 81.21.33.35:5060 -> pstngw:39871 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK01c64e769ba6c176. From: "12345"<sip:12345@81.21.33.35:5060>;tag=9ca5c1fe3b43aad6. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E6010EC-1764. Date: Wed, 06 Sep 2006 10:27:40 GMT. Call-ID: 72d0e4e8adca2ab2(a)192.168.1.102. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 32352 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 237. . v=0. o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw s=SIP Call. c=IN IP4 81.21.33.35.-------------------------------------<in pstngw it noticed the contact header of SIP-SERVER>--------------------- t=0 0. m=audio 60518 RTP/AVP 18 100. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. a=ptime:20. ---------------------------------------------<after talking some time if any of UA hung the phone (here i am showing from pstn hung up) the result goes like this---------------- U pstngw:52991 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on SIP/2.0. Via: SIP/2.0/UDP 81.21.38.15:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 6. Route: <sip:192.168.1.100:5060>, <sip:82.102.69.105:39871>.------------------------> What is happening here ??????? Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. . # U 81.21.33.35:5060 -> 192.168.1.100:5060--------------------------------------------------->here unexpected problem arises for me it have to use 82.102.69.105 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=7E66E3D8-1292;lr=on>. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bKb0f1.3fe32691.0. Via: SIP/2.0/UDP pstngw:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 5. Route: <sip:82.102.69.105:39871>. Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. P-hint: rr-enforced. So the call never ends up from SIP SERVER to other party Make some comments on above and assist me where am I going wrong: and i am using www.openser.org pstn default script as ser.cfg Hope to get some help Thanks, Ravi.

1 0

[Serusers] SER & Asterisk (IVR problem)
by Roberto Lopes 06 Sep '06

06 Sep '06

Hi to all! I've a big problem with SER & Asterisk I've a network where i put them working together. Asterisk is the media gateway with PSTN as well as being the server which is making all the services available (IVR, voicemail, conference calls, etc). SER just routes calls either to the softphones, VoIP phones or to Asterisk who will just forward the calls to PSTN (via Asterisk). Till now everything works well except the service IVR which is working on Asterisk. Details: 192.168.226.13 - IP SER 192.168.226.45 - IP Asterisk The problem is only after the user (roberto(a)192.168.226.13 - authenticated and authorized in RADIUS) calls IVR. Till then the user calls, IVR gives some information to the user and everything goes well. Problem is when he presses a number which will have to redirect the call to another user. And why is there a problem. Because the initially roberto(a)192.168.226.13 to be able to call IVR had to rewrite host & port 192.168.226.45 so that the call would reach Asterisk who would then call to the IVR. That rewriting turned roberto(a)192.168.226.13 into roberto(a)192.168.226.45. That is, I can say, that it became a "new user". While it's on the IVR there's no problem, but when call is diverted it has to give its credentials... Credentials that don't exist in RADIUS cause it only accepts usernames that end in @192.168.226.13 (IP of SER) Here is the following code, both of extensions file in Asterisk and configure file of SER I hope somebody knows how to solve it or at least find a way how to. extensions.conf - Asterisk [sip] exten => 74001,1,Answer exten => 74001,2,SetMusicOnHold(default) exten => 74001,3,Set(TIMEOUT(digit)=5) exten => 74001,4,Set(TIMEOUT(response)=10) exten => 74001,5,Background(menu) exten => 1,1,Dial(SIP/roberto@192.168.226.13:5060,,r) ... exten => 7,1,Goto(sip,74001,1) exten => 8,1,Hangup ser.cfg - SER ... if ((uri=~"^sip:7[0-9]{4}@.*")) { route(5); break; }; ... route[5] { # Redirecting to Asterisk # MsgLog xlog("L_ALERT","(WARNING) Sending to Asterisk request of %fu to %tu\n"); rewritehostport("192.168.226.45:5060"); route(1); } Thanks for your help, Roberto Lopes

2 1

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path (from NAT)there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;-http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/…<http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin…> Regards, Ravi.

1 0

[Serusers] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

2 3

Re: [Users] more on accounting
by tele 06 Sep '06

06 Sep '06

Glenn Dalgliesh wrote: >This is from previous list > >"i Glenn, > >For this feature to work, you need to be sure that you do record_route() on >the INVITE and that "append_fromtag" RR module param is not disable. > >regards, >bogdan " > > Yes i have read that thread. i always record_route all messages before do things. if (!method=="REGISTER") record_route(); by the sample openser.cfg and for be sure i have also forced: modparam("rr","append_fromtag",1) >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf >Of tele >Sent: Wednesday, June 28, 2006 11:05 AM >To: users(a)openser.org >Subject: [Users] more on accounting > >Hi all, > >I'm using OpenSER v1.0.1 and i have done an accounting compatible with >my billing system. >Now i have problem with detection of flow direction for billing reason. > >i'm talking about openser 1.0.x: > >the problem is when the BYE come from callee. >i know the problem is solved in openser1.1.x with the > >modparam("acc", "detect_direction", 1) > >i have read in modules documentation of 1.0.x that i can use >is_direction() of RR module for detect call flow direction but it says >that this must be called after loose_route() > >so for the accounting i must call setflag before loose_route() but what >can i do if i want detect call flow >before of exec setflag? > >Hmmm i think that i must install OpenSER 1.1.x and test with it > >:tele > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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sr-users September 2006