sr-users July 2003

sr-users@lists.kamailio.org

67 participants
160 discussions

Re: [Serusers] rewrite & ACK forwarding problem
by lgfausak＠august.net 30 Apr '04

30 Apr '04

I have also had problems with getting the ACK back. I don't completely understand your configuration, you must allow for packets going both directions, right? Here is my config : route { # check to see if the message has been around too long # probably means that it is looping # if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; # # make sure the length of the message isn't too long! # if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; # # do the loose-routing thing, this is important! # if(loose_route()) { log(1,"doing top loose route"); t_relay(); break; }; # this is where I was dropping the ACKS. # I was simply dropping these, but they must be relayed # because they can be ACKs if(!(uri==myself)) { if(!t_relay()) { sl_reply_error(); break; }; break; }; This gets the ACKs through for me. By the way, I have this configured with Cisco ATAs, version 2.16. ---greg > >I have the same problem and posed it to the group yesterday ([Serusers] >Ignored 200 OK message.) So far the only workaround that I have found is to >use the rules in my gateway to rewrite the dialed digits before sending them >to the PSTN PRI, thus leaving the origianl URI intact for SIP >communications. > >One person told me that this is a bug in the Cisco ATA, but it happens on my >IPDialog phones also. It seems to me that the INVITE is being processed by >the SER dial rules and is rewritten, but the ACK is not. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/ > >----- Original Message ----- >From: "Alexander Mayrhofer" <axelm(a)nic.at> >To: <serusers(a)lists.iptel.org> >Sent: Friday, June 27, 2003 12:15 PM >Subject: [Serusers] rewrite & ACK forwarding problem > > >> >> Hi, >> >> we're running SER together with a PSTN Gateway. Before a call get's >> forwarded to the gateway, we are rewriting the request URI to make >> rewriting on the GW as simple as possible: >> >> route { >> ... >> strip(3); # +43xxx -> xxx >> prefix("0"); # xxx -> 0xxx >> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway >> route(1); >> break; >> ... >> >> SIP call flow looks like (record route enabled): >> >> (1) phone -> SER >> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0 >> >> (2) SER -> phone >> SIP/2.0 100 trying -- your call is important to us >> >> (3) SER -> GW >> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> (4) GW -> SER >> SIP/2.0 100 Trying >> >> (5) GW -> SER >> SIP/2.0 183 Session Progress >> >> (6) SER -> phone >> SIP/2.0 183 Session Progress >> >> (7) GW -> SER >> SIP/2.0 180 Ringing >> >> (8) SER -> phone >> SIP/2.0 180 Ringing >> >> (9) GW -> SER >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060> >> >> (10) SER -> phone >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060> >> >> [ call established, we can talk, but ... ] >> >> (11) phone -> SER >> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> --> Here starts the problem. That ACK (11) never gets forwarded to the >> Gateway, so after a few seconds, the GW starts over at (9). Those three >> packets (9-11) repeat a few times until GW runs into a timeout and drops >> the call. >> >> I have the impression that SER can't match the packet to the previous >> requests because of the rewritten URI. Is that correct? >> >> The only output at debug level 3 is: >> >> Warning: sl_send_reply: I won't send a reply for ACK!! >> >> Is that a routing goof somewhere in our scripts or is that a more >> generic problem? Is the problem that the warning indicates somehow >> related to the fact that the ACK is not being forwarded? >> >> Help appreciated. >> >> cheers >> >> axelm >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Xten and SER problem
by Jesus Rodriguez 17 Nov '03

17 Nov '03

Hello, I'm trying to register an Xten (1016 build) to SER and i always get a "401" response. Registering an ATA186 to the same SER with the same user and pass works with no problem (logs below Xten). We have tried SIPPS and registers ok too. Xten send the REGISTER, SER answers with a WWW-Authenticate challenge but Xten doesn't send anything, just sends a new REGISTER after few seconds: (c)2003 Xten Networks Inc. All rights reserved. Private build: 1016 SIP: 192.168.1.203:5060 RTP: 192.168.1.203:8000 NAT: 192.168.1.203 PXY#0: 192.168.1.2:5060 SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562200000000e06a238bd1858769f2e46b11d8d837f1",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" The same action works ok with ATA186: [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c579a00000000aa6fd2432e429f54e3 7e75981218834b",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13590 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" [0]Reg Resp 401; Unauthorized [0]REGISTER Retry 0 [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Authorization: Digest username="4070",realm="voztelecom.net",nonce="3f1c579a0000 0000aa6fd2432e429f54e37e75981218834b",uri="sip:192.168.1.2",response="fe07b8be83 0305cca7a28bf35c112270" Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;q=0.00;expires=6 0 Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" Thanks in advance. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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[Serusers] 45% OFF TOPIC (SIPTel settings)
by Director General: NEFACOMP 25 Aug '03

25 Aug '03

I have recently bought a SIPTel device (http://www.sipmicro.com) but after reading, rereading and rereading the manuals and the webpages I can't figure out how to use my phone with SER because I can't see where to set the SIP proxy address. Does any one know how I can do this? I sent an Email to them asking but they have never replied. The one who sold it to me is not willing or is not well trained to know how to tell me that because she seems to not know it. Of course it works with SIP proxy server of the one who sold it to me. Thanks, __________________________________ NZEYIMANA Emery Fabrice NEFA Computing Services, Inc. P.O. Box 5078 Kigali Office Phone: +250-51 11 06 Office Fax: +250-50 15 19 Mobile: +250-08517768 Email: dg(a)nefacomp.net http://www.nefacomp.net/

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Re: [Serusers] ser support IPv6 ?
by Jiri Kuthan 25 Aug '03

25 Aug '03

We have used our own very prototypical one, bonephone. -jiri At 06:18 AM 7/19/2003, Ng, Soo Sim wrote: >May I know which IPv6 UA Client do you use? >SSng > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Problems and doubts
by giAndrea 04 Aug '03

04 Aug '03

Hi to all i've some doubts: I've installed SER on a Linux RedHad 9.0. 1 - Why I can't see another SER server on a lan with MSN? Ex. andrea(a)foo.bar cannot send message and view andrea(a)foo2.bar 2 - On seruser.pdf at page 50 i've read that "The latest SER release includes automated multidomain management which greatly automates maintenance of multiple domains.". I've seen on mySql db that there is the domain table with domain primary key and on subscriber table user_name and domain are primary key. I think that there is exist a foreign key from subscriber to domain. Who has to write the information on this table? How do i configure ser to use multiple domain? 3 - How do I configure ser to report information on mySql database? For example, i've configured ser for Reporting Missed Calls by the acc module but there is an error on acc_request("404 Not Found"); instruction on ser.cfg. 4 - Microsoft MSN 6.0 doesn't support service communication. Is it true? 5 - On seruser.pdf at page 87 there are the old version of database tables. 6 - Where can i found help for ser command line options? regards, Andrea

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[Serusers] mysql auth query
by Madan 04 Aug '03

04 Aug '03

hi all how can one change the mysql query which does the auth as per the individual specific requirements? regards, Madan

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RE: [Serusers] Does lookup("location") queries the database or the memory?
by Klaus Darilion 01 Aug '03

01 Aug '03

Ok, so I will try another setup: 2 proxies, each of them interacts with a separate mysql database. Registrations will be replicated between the proxies using t_replicate. I have also seen a replication parameter in serweb when adding new contacts via the web interface and the FIFO. I found no documentation about it, but I guess this will also replicate the new contact to another sip proxy (don't know if its implemented yet). So, both proxies can be used for registration and normal proxy tasks. But how can a proxy catch up the lost REGISTER messages from the other proxy during a "down" period? I think, one possibility would be, that when the proxy/database is "up" again, the database begins to mirror the complete state from the other database, and only when this is finished, ser will be started again and reads the current user location from the database. Is there another, maybe simpler solution to this problem? Klaus > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: Wednesday, July 30, 2003 1:43 PM > To: Klaus Darilion > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Does lookup("location") queries the > database or the memory? > > > Hello, > > On 30-07 13:39, Klaus Darilion wrote: > > Hello! > > > > I have found the db_mode option in the userloc module, > which controls > > updates in the location table. But how are the lookups > handled in ser? > > > > If a user location will be requested by lookup("location"), > will this query > > the backend database or only the RAM. Is this lookup behaviour also > > configurable (how?)? > > No, it will not query the database, only memory will be used. > > > I'm thinking about a configuration with 2 ser proxies and > one database > > backend. If the lookup only queries the cached location > table, I would have > > to froward the registrations between the two proxies via > t_replicate. But > > wouldn't this cause both proxies to update the database > resulting in two > > entries in the location table for each registration? > > Yes, this is not possible at the moment. Another database mode which > will query the backend database when doing lookups is on > our todo list. > > Jan. > >

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[Serusers] UAs' interoperability with SER
by George KAPELIOS 31 Jul '03

31 Jul '03

Using SER with Xlite softphone (either v.1.x or 2.0) through the Internet, one-way communication is experienced by the UA (only outgoing traffic is possible). The UAs are connected to the Internet via public IP address and no firewall is enabled. The problem has been replicated with 3 different users, over 3 different ISPs. However, the same users, can always communicate without problem over the same connections, using Windows Messenger. In both cases, I have captured (ngrep) the SIP packets from the server port (see here after). The topology of the SER and UAs, is as following: UA1 -- public IP ---- Internet ---- NAT -- ATA186 (Xlite) (dialup) | (DSL) or | (Win Mes) | | Router(NAT) | DMZ | SER (listen on 5070 udp) Could anyone take a look in the packets for Xlite and Messenger and see what makes the first one acting as being behind a firewall, while Win Messenger is working properly ??? Thanks, George PS. Xlite is working properly with the SER in Intranet environment. =================================================== Windows Messenger =================== U 62.4.193.25:3587 -> 10.0.0.9:5070 REGISTER sip:voiceglobe.net SIP/2.0.. Via: SIP/2.0/UDP 62.4.193.25:10616.. From: <sip:kapelios@voiceglobe.net>;tag=8c67d569-eac2-4665-89bd-a800e64c9f52.. To: <sip:kapelios@voiceglobe.net>.. Call-ID: 9dc07f3f-4c48-48e4-8d56-a77b8dc5a044(a)62.4.193.25.. CSeq: 1 REGISTER.. Contact: <sip:62.4.193.25:10616>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK".. User-Agent: Windows RTC/1.0.. Expires: 1200.. Event: registration.. Allow-Events: presence.. Content-Length: 0.... # U 10.0.0.9:5070 -> 62.4.193.25:10616 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 62.4.193.25:10616.. From: <sip:kapelios@voiceglobe.net>;tag=3efd91fa-22c4-4c78-bc54-8896cb8a4617.. To: <sip:kapelios@voiceglobe.net>;tag=46a6e639fa02362ac1ba4fea686e961.7623.. Call-ID: c8a90fea-69c9-4eb0-b0fa-46bcf1db4253(a)62.4.193.25.. CSeq: 2 REGISTER.. Contact: <sip:62.4.193.25:10616>;q=0.00;expires=1200.. Server: Sip EXpress router (0.8.11pre29 (i386/linux)).. Content-Length: 0.. Warning: 392 10.0.0.9:5070 "Noisy feedback tells: pid=1155 req_src_ip=62.4.193.25 req_src_port=3588 in_uri=sip:voiceglobe.net out_uri=sip:voiceglobe.net via_cnt==1".... # U 62.4.193.25:3588 -> 10.0.0.9:5070 INVITE sip:11111@voiceglobe.net SIP/2.0.. Via: SIP/2.0/UDP 62.4.193.25:10616.. From: "kapelios" <sip:kapelios@voiceglobe.net>;tag=2f2be407-7d38-4dd0-938b-068517646fd0.. To: <sip:11111@voiceglobe.net>.. Call-ID: 54b641af-9616-481c-8e37-cb09797a266a(a)62.4.193.25.. CSeq: 1 INVITE.. Contact: <sip:62.4.193.25:10616>.. User-Agent: Windows RTC/1.0.. Content-Type: application/sdp.. Content-Length: 293.... v=0.. o=i8200 0 0 IN IP4 62.4.193.25.. s=session.. c=IN IP4 62.4.193.25.. b=CT:32.. t=0 0.. m=audio 37890 RTP/AVP 97 111 4 3 101.. a=rtpmap:97 red/8000.. a=rtpmap:111 SIREN/16000.. a=fmtp:111 bitrate=16000.. a=rtpmap:4 G723/8000.. a=rtpmap:3 GSM/8000.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-16.. # U 10.0.0.9:5070 -> 62.4.193.25:10616 SIP/2.0 100 trying -- your call is important to us.. Via: SIP/2.0/UDP 62.4.193.25:10616.. From: "kapelios" <sip:kapelios@voiceglobe.net>;tag=2f2be407-7d38-4dd0-938b-068517646fd0.. To: <sip:11111@voiceglobe.net>.. Call-ID: 54b641af-9616-481c-8e37-cb09797a266a(a)62.4.193.25.. CSeq: 1 INVITE.. Server: Sip EXpress router (0.8.11pre29 (i386/linux)).. Content-Length: 0.. Warning: 392 10.0.0.9:5070 "Noisy feedback tells: pid=1156 req_src_ip=62.4.193.25 req_src_port=3588 in_uri=sip:11111@voiceglobe.net out_uri=sip:autoanswer@68.212.243.205:5060;transport=udp via_cnt==1".... # U 10.0.0.9:5070 -> 68.212.243.205:5060 INVITE sip:autoanswer@68.212.243.205:5060;transport=udp SIP/2.0.. Max-Forwards: 10.. Record-Route: <sip:11111@10.0.0.9:5070; ftag=2f2be407-7d38-4dd0-938b-068517646fd0;lr>.. Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK6adc.a29540c4.0.. Via: SIP/2.0/UDP 62.4.193.25:10616.. From: "kapelios" <sip:kapelios@voiceglobe.net>;tag=2f2be407-7d38-4dd0-938b-068517646fd0.. To: <sip:11111@voiceglobe.net>.. Call-ID: 54b641af-9616-481c-8e37-cb09797a266a(a)62.4.193.25.. CSeq: 1 INVITE.. Contact: <sip:62.4.193.25:10616>.. User-Agent: Windows RTC/1.0.. Content-Type: application/sdp.. Content-Length: 293.. .. v=0..o=i8200 0 0 IN IP4 62.4.193.25.. s=session.. c=IN IP4 62.4.193.25.. b=CT:32.. t=0 0.. m=audio 37890 RTP/AVP 97 111 4 3 101.. a=rtpmap:97 red/8000.. a=rtpmap:111 SIREN/16000.. a=fmtp:111 bitrate=16000.. a=rtpmap:4 G723/8000.. a=rtpmap:3 GSM/8000.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-16.. # U 68.212.243.205:5060 -> 10.0.0.9:5070 SIP/2.0 180 Ringing.. Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK6adc.a29540c4.0.. Via: SIP/2.0/UDP 62.4.193.25:10616.. From: "kapelios" <sip:kapelios@voiceglobe.net>;tag=2f2be407-7d38-4dd0-938b-068517646fd0.. To: <sip:11111@voiceglobe.net>;tag=2796180519.. Call-ID: 54b641af-9616-481c-8e37-cb09797a266a(a)62.4.193.25.. CSeq: 1 INVITE.. Server: Cisco ATA 186 v2.16 ata18x (030509f1).. Content-Length: 0.... # U 68.212.243.205:5060 -> 10.0.0.9:5070 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK6adc.a29540c4.0.. Via: SIP/2.0/UDP 62.4.193.25:10616.. Record-Route: <sip:11111@10.0.0.9:5070;ftag=2f2be407-7d38-4dd0-938b-068517646fd0;lr>.. From: "kapelios" <sip:kapelios@voiceglobe.net>;tag=2f2be407-7d38-4dd0-938b-068517646fd0.. To: <sip:11111@voiceglobe.net>;tag=2796180519.. Call-ID: 54b641af-9616-481c-8e37-cb09797a266a(a)62.4.193.25.. CSeq: 1 INVITE.. Contact: <sip:autoanswer@68.212.243.205:5060;transport=udp>.. Server: Cisco ATA 186 v2.16 ata18x (030509f1).. Content-Length: 201.. Content-Type: application/sdp.. .. v=0.. o=autoanswer 4767 4767 IN IP4 192.168.1.50.. s=ATA186 Call.. c=IN IP4 68.212.243.205.. t=0 0.. m=audio 20002 RTP/AVP 4 101.. a=rtpmap:4 G723/8000/1.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-15.. # ================================================== X-Lite ======== U 62.4.193.25:5060 -> 10.0.0.9:5070 REGISTER sip:voiceglobe.net SIP/2.0.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: <sip:nikoline@voiceglobe.net>.. To: <sip:nikoline@voiceglobe.net>.. Contact: "G. Kapelios" <sip:nikoline@62.4.193.25:5060>.. Call-ID: 1F385A1BD0C648B381836793799E029A(a)voiceglobe.net.. CSeq: 16149 REGISTER.. Expires: 500.. User-Agent: X-Lite build 1011.. Content-Length: 0.... # U 10.0.0.9:5070 -> 62.4.193.25:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: <sip:nikoline@voiceglobe.net>.. To: <sip:nikoline@voiceglobe.net>; tag=46a6e639fa023622ac1ba4fea686e961.7d05.. Call-ID: 1F385A1BD0C648B381836793799E029A(a)voiceglobe.net.. CSeq: 16150 REGISTER.. Contact: <sip:nikoline@62.4.193.25:5060>;q=0.00;expires=500.. Server: Sip EXpress router (0.8.11pre29(i386/linux)).. Content-Length: 0.. Warning: 392 10.0.0.9:5070 "Noisy feedback tells: pid=1156 req_src_ip=62.4.193.25 req_src_port=5060 in_uri=sip:voiceglobe.net out_uri=sip:voiceglobe.net via_cnt==1".... # U 62.4.193.25:5060 -> 10.0.0.9:5070 INVITE sip:11111@voiceglobe.net SIP/2.0.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: G. Kapelios <sip:nikoline@voiceglobe.net>;tag=301448250.. To: <sip:11111@voiceglobe.net>.. Contact: <sip:nikoline@62.4.193.25:5060>.. Call-ID: 7BCE0A51-7CC0-4892-BB6E-F10E67CFA94E(a)62.4.193.25.. CSeq: 18469 INVITE.. Content-Type: application/sdp.. User-Agent: X-Lite build 1011.. Content-Length: 290.. .. v=0.. o=nikoline 22941408 22941408 IN IP 4 62.4.193.25.. s=X-Lite.. c=IN IP4 62.4.193.25.. t=0 0.. m=audio 8000 RTP/AVP 0 8 3 98 97 101.. a=rtpmap:0 pcmu/8000.. a=rtpmap:8 pcma/8000.. a=rtpmap:3 gsm/8000.. a=rtpmap:98 iLBC.. a=rtpmap:97 speex/8000.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-15.. # U 10.0.0.9:5070 -> 62.4.193.25:5060 SIP/2.0 100 trying -- your call is important to us.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: G. Kapelios <sip:nikoline@voiceglobe.net>;tag=301448250.. To: <sip:11111@voiceglobe.net>.. Call-ID: 7BCE0A51-7CC0-4892-BB6E-F10E67CFA94E(a)62.4.193.25.. CSeq: 18469 INVITE.. Server: Sip EXpress router (0.8.11pre29 (i386/linux)).. Content-Length: 0.. Warning: 392 10.0.0.9:5070 "Noisy feedback tells: pid=1155 req_src_ip=62.4.193.25 req_src_port=5060 in_uri=sip:11111@voiceglobe.net out_uri=sip:autoanswer@68.212.243.205:5060;transport=udp via_cnt==1".. # U 10.0.0.9:5070 -> 68.212.243.205:5060 INVITE sip:autoanswer@68.212.243.205:5060;transport=udp SIP/2.0.. Max-Forwards: 10..Record-Route: <sip:11111@10.0.0.9:5070;ftag=301448250;lr>.. Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK1882.4c4fc6e3.0.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: G. Kapelios <sip:nikoline@voiceglobe.net>;tag=301448250.. To: <sip:11111@voiceglobe.net>.. Contact: <sip:nikoline@62.4.193.25:5060>.. Call-ID: 7BCE0A51-7CC0-4892-BB6E-F10E67CFA94E(a)62.4.193.25.. CSeq: 18469 INVITE.. Content-Type: application/sdp..User-Agent: X-Lite build 1011.. Content-Length: 290.. .. v=0.. o=nikoline 22941408 22941408 IN IP4 62.4.193.25.. s=X-Lite..c=IN IP4 62.4.193.25.. t=0 0.. m=audio 8000 RTP/AVP 0 8 3 98 97 101.. a=rtpmap:0 pcmu/8000.. a=rtpmap:8 pcma/8000.. a=rtpmap:3 gsm/8000.. a=rtpmap:98 iLBC.. a=rtpmap:97 speex/8000.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-15.. # U 68.212.243.205:5060 -> 10.0.0.9:5070 SIP/2.0 100 Trying.. Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK1882.4c4fc6e3.0.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: G. Kapelios <sip:nikoline@voiceglobe.net>;tag=301448250.. To: <sip:11111@voiceglobe.net>;tag=1719463769.. Call-ID: 7BCE0A51-7CC0-4892-BB6E-F10E67CFA94E(a)62.4.193.25.. CSeq: 18469 INVITE.. Server: Cisco ATA 186 v2.16 ata18x (030509f1).. Content-Length: 0.. # U 68.212.243.205:5060 -> 10.0.0.9:5070 SIP/2.0 180 Ringing.. Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK1882.4c4fc6e3.0.. Via: SIP/2.0/UDP 62.4.193.25:5060.. From: G. Kapelios <sip:nikoline@voiceglobe.net>;tag=301448250.. To: <sip:11111@voiceglobe.net>;tag=1719463769.. Call-ID: 7BCE0A51-7CC0-4892-BB6E-F10E67CFA94E(a)62.4.193.25.. CSeq: 18469 INVITE..Server: Cisco ATA 186 v2.16 ata18x (030509f1).. Content-Length: 0.. # U 68.212.243.205:5060 -> 10.0.0.9:5070 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 10.0.0.9:5070;branch=z9hG4bK1882.4c4fc6e3.0.. Via: SIP/2.0/UDP 62.4.193.25:5060.. Record-Route: <sip:11111@10.0.0.9:5070;ftag=301448250;lr>.. From: G. Kapelios <sip:nikoline@voiceglobe.net>;tag=301448250.. To: <sip:11111@voiceglobe.net>;tag=1719463769.. Call-ID: 7BCE0A51-7CC0-4892-BB6E-F10E67CFA94E(a)62.4.193.25.. CSeq: 18469 INVITE.. Contact: <sip:autoanswer@68.212.243.205:5060;transport=udp>.. Server: Cisco ATA 186 v2.16 ata18x (030509f1).. Content-Length: 201.. Content-Type: application/sdp.. .. v=0.. o=autoanswer 5988 5988 IN IP4 192.168.1.50.. s=ATA186 Call..c=IN IP4 68.212.243.205.. t=0 0..m=audio 20002 RTP/AVP 0 101.. a=rtpmap:0 PCMU/8000/1.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-15.. #

2 2

[Serusers] Checking user credentials on INVITE method
by Director General: NEFACOMP 31 Jul '03

31 Jul '03

I know Jan won't be happy for this question because it may look simple. But I have tried everything I can and after failing I decided to ask it here. I have realized that SER checks the username and password when a UA sends a REGISTER command but it doesn't check the password when a user sends a INVITE command. This makes SER accept calls from people who did not even register. This is true because my GATEWAY doesn't register with SER but I can make calls from PSTN to my IP network. So, my question is: HOW do I instruct SER to check the username and password of the CALLER? Thanks, __________________________________ NZEYIMANA Emery Fabrice NEFA Computing Services, Inc. P.O. Box 5078 Kigali Office Phone: +250-51 11 06 Office Fax: +250-50 15 19 Mobile: +250-08517768 Email: dg(a)nefacomp.net http://www.nefacomp.net/

3 3

[Serusers] Difference <> proxy_authorize and www_authorize
by Director General: NEFACOMP 31 Jul '03

31 Jul '03

Can someone tell me difference between proxy_authorize() and www_authorize() functions? Thanks, __________________________________ NZEYIMANA Emery Fabrice NEFA Computing Services, Inc. P.O. Box 5078 Kigali Office Phone: +250-51 11 06 Office Fax: +250-50 15 19 Mobile: +250-08517768 Email: dg(a)nefacomp.net http://www.nefacomp.net/

2 1

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sr-users July 2003