sr-users February 2004

sr-users@lists.kamailio.org

144 participants
382 discussions

Re: [Serusers] Password
by Jiri Kuthan 12 Feb '04

12 Feb '04

At 02:11 AM 2/10/2004, kapil dhawan wrote: >thats absolutely fine i know md5 is used....i can also compare coming password in requested with my md5 converted password but i am stuck with how and where to read pasword information coming in a request The authentication module does it for you. It takes the MD5-hashed value from request (example bellow) and compares it against expected value based on credentials stored in subscriber table. The table may be located in mysql, dbtext or whereever. -jiri example: Authorization: DIGEST username="17479383213", realm="proxy01.foobar.com", algorithm=MD5, uri="sip:proxy01.foobar.com", nonce="4028321e2577ba09167c3c5702bc2105d560dbaa", response="6e71e5747df3198d24d00fb2a8733392" >>From: Jiri Kuthan <jiri(a)iptel.org> >>To: "kapil dhawan" <oswriter(a)hotmail.com>, serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Password >>Date: Mon, 09 Feb 2004 08:27:26 +0100 >> >>At 05:04 AM 2/9/2004, kapil dhawan wrote: >>>Hi >>>How can i retrieve password coming in 'REGISTER' Request. >> >>There is fortunately no easy way to do it. SIP used MD5 digest >>function which would be quite hard to reverse for you. >> >>>problem is i don't want to do mysql authentication as i have less space... >> >>use dbtext module. >> >>-jiri > >_________________________________________________________________ >Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Non-answered transmission by CPL
by Koichi Taniguchi 12 Feb '04

12 Feb '04

Hello. I have succeeded in operation of CPL function. However, there is a script which does not operate in the following cases. <Non-answered transmission by CPL> incoming-> location url -> proxy timeout="10"-> noanswer-> proxy -- You need to point out the problem of my script ---------------------------------------------------- <?xml version="1.0"?> <!DOCTYPE cpl PUBLIC "-//IETF//DTD RFCxxxx CPL 1.0//EN" "cpl-06.dtd"> <cpl> <incoming> <location url="sip:1000@192.168.100.11:5060"> <proxy timeout="10"> <noanswer> <location url="sip:2000@192.168.100.11:5060"> <location url="sip:3000@192.168.100.11:5060"> <proxy ordering="parallel" /> </location> </location> </noanswer> </proxy> </location> </incoming> </cpl> ------------------------------------------------------ Taniguchi / PanasonicCC

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[Serusers] sip messages
by Annie Sasidar 12 Feb '04

12 Feb '04

Hi, Attached with this email is the ethereal o/p for communication that happens between two wince clients registering on ser using TCP. My problem is that once conversation starts between the two clients within moments a network error occurs and the converstaion stops. I am trying to trace the SIP communication between the server and the client to see if there is any clue as to why the communication terminates all of a sudden. Experts please help me with your insight on this. Thanks, Annie

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[Serusers] auth module - append_rpid_hf
by Gavin Bensom 12 Feb '04

12 Feb '04

I'm interested in using the append_rpid_hf function. The auth module doc states that the function inserts "the saved value of the SIP URI received from the database or radius server followed by . . . " I've inserted the function into my config file, plus carefully inserted the necessary loadmodule and modparam statements but no RPID field is being created [based on ethereal captures at the gateway]. I think that its because I'm not doing the right thing to get a SIP URI saved to memory. I'm not sure what I'm missing here. Also, my subscriber table has no RPID column in it so I pointed the rpid fuction to the phone column. Also, will the radius module work with any [i.e. non linux/unix] radius server? I'm running 0.8.12 on RH 9 version: ser 0.8.12-tcp_nonb (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.1 2003/11/26 23:45:50 andrei Exp $ main.c compiled on 12:58:28 Jan 2 2004 with gcc 3.2 Thanks, G. __________________________________ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html

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Re: [Serusers] Rewrite URI
by kapil dhawan 12 Feb '04

12 Feb '04

thx we got it in action.c and implemented it also.. Now there is one more issue i like to discuss...... what we are doing is...have written a module and every INVITE request we are forwarding it to that place...... checks done there 1. is call allowed 2. is call denied 3. has call to be forwarded...etc and few more.... Now if we are forwarding it to some number so in ser.cfg we just relay the call after writing new uri into the structure.....now if we want to do the same checks of allow , deny and forward etc on the same call....will recursion do it in our module or we have to play with ser.cfg..... regards >From: Jan Janak <jan(a)iptel.org> >To: kapil dhawan <oswriter(a)hotmail.com> >CC: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Rewrite URI >Date: Wed, 11 Feb 2004 20:19:34 +0100 > >See rewrite_RURI function in sip_router/modules/rr/loose.c > > Jan. > >On 11-02 09:27, kapil dhawan wrote: > > I want to rewrite URI in one of my modules. How can i do it....i know to >do > > it in ser.cfg file but not in my module..... > > > > _________________________________________________________________ > > Easiest Money Transfer to India. Send Money To 6000 Indian Towns. > > http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Masterpieces made affordable! Buy art prints. http://go.msnserver.com/IN/42736.asp MSN Shopping.

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[Serusers] One way voice Problem
by Balaji Bapulal Thoguluva 12 Feb '04

12 Feb '04

Hi, I have SER and 2 cisco SIP phones configured. When I call between the SIP phones, I can hear voice from only direction (callee to caller voice works, caller to callee doesn't work). Both phones display connected in their LCD screen and I also checked the codecs used by the phones are same. I would appreciate if anyone can point me where this gets wrong and what are the possible solutions for this problem. I have attached the ngrep output below. In the second message of the output, I can see a "Warning: 392" header field with a value "Noisy Feedback tells:...". ngrep output: ------------ interface: eth0 (192.168.6.0/255.255.255.0) ############# # U 192.168.6.103:49834 -> 192.168.6.100:5060 INVITE sip:3311@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b 001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>..Call-ID: 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Date: Wed, 11 Feb 2 004 18:33:50 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip:3322@192.168 .6.103:5060..Content-Type: application/sdp..Content-Length: 226....v=0..o=CiscoSystemsSIP-IPPhone-UserAgent 26498 14989 IN IP4 192.168 .6.103..s=SIP Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 27234 RTP/AVP 0 8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-even t/8000..a=fmtp:101 0-11.. # U 192.168.6.100:5060 -> 192.168.6.103:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.10 0>;tag=925f0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>..Call-ID: 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..CSeq: 101 INVITE..Server: Sip EXpress router (0.8.11 (i386/linux))..Content-Length: 0..Warning: 392 192.168.6.100:5060 "Noisy feedback tells : pid=9732 req_src_ip=192.168.6.103 req_src_port=49834 in_uri=sip:3311@192.168.6.100 out_uri=sip:3311@192.168.6.104:5060 via_cnt==1". ... # U 192.168.6.100:5060 -> 192.168.6.104:5060 INVITE sip:3311@192.168.6.104:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:3311@192.168.6.100;ftag=925f0b001c0632395f2e8-34ee90e ;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch=z9hG4bK8153.9c3bbf61.0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <si p:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>..Call-ID: 000b5f92-63c00003-63cc121e-248e7823@19 2.168.6.103..Date: Wed, 11 Feb 2004 18:33:50 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: applicatio n/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Type: application/sdp..Content-Length: 226....v=0..o=CiscoSystemsSIP-IPPhone-User Agent 26498 14989 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 27234 RTP/AVP 0 8 18 101..a=rtpmap:0 pcmu/8 000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11.. # U 192.168.6.104:50020 -> 192.168.6.100:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.6.100;branch=z9hG4bK8153.9c3bbf61.0,SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji " <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-ID : 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Server: Cisco-SIP-IP-Phone/2..Date: Wed, 11 Feb 2004 18:33:50 GMT..CSeq: 101 INVI TE..Content-Length: 0.... # U 192.168.6.104:50020 -> 192.168.6.100:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.6.100;branch=z9hG4bK8153.9c3bbf61.0,SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaj i" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-I D: 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Server: Cisco-SIP-IP-Phone/2..Date: Wed, 11 Feb 2004 18:33:50 GMT..CSeq: 101 INV ITE..Content-Length: 0.... # U 192.168.6.100:5060 -> 192.168.6.103:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34 ee90e..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-ID: 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Se rver: Cisco-SIP-IP-Phone/2..Date: Wed, 11 Feb 2004 18:33:50 GMT..CSeq: 101 INVITE..Content-Length: 0.... # U 192.168.6.104:50020 -> 192.168.6.100:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.6.100;branch=z9hG4bK8153.9c3bbf61.0,SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <s ip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-ID: 00 0b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Server: Cisco-SIP-IP-Phone/2..Contact: sip:3311@192.168.6.104:5060..Record-Route: <si p:3311@192.168.6.100;ftag=925f0b001c0632395f2e8-34ee90e;lr=on>..Date: Wed, 11 Feb 2004 18:33:53 GMT..CSeq: 101 INVITE..Content-Type: a pplication/sdp..Content-Length: 221....v=0..o=CiscoSystemsSIP-IPPhone-UserAgent 14538 26512 IN IP4 192.168.6.104..s=SIP Call..c=IN IP4 192.168.6.104..t=0 0..m=audio 24138 RTP/AVP 0 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11.. # U 192.168.6.100:5060 -> 192.168.6.103:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e ..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-ID: 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Server: Cisco-SIP-IP-Phone/2..Contact: sip:3311@192.168.6.104:5060..Record-Route: <sip:3311@192.168.6.100;ftag=925f0b001c0632395f2e8-34ee90e; lr=on>..Date: Wed, 11 Feb 2004 18:33:53 GMT..CSeq: 101 INVITE..Content-Type: application/sdp..Content-Length: 221....v=0..o=CiscoSyste msSIP-IPPhone-UserAgent 14538 26512 IN IP4 192.168.6.104..s=SIP Call..c=IN IP4 192.168.6.104..t=0 0..m=audio 24138 RTP/AVP 0 101..a=rt pmap:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..U 192.168.6.103:49834 -> 192.168.6.100:5060 ACK sip:3311@192.168.6.100:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f 0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-ID: 000b5f92-63c00003-63cc121e-248e78 23@192.168.6.103..Date: Wed, 11 Feb 2004 18:33:53 GMT..Route: <sip:3311@192.168.6.104:5060>..CSeq: 101 ACK..Content-Length: 0.... # U 192.168.6.100:5060 -> 192.168.6.104:5060 ACK sip:3311@192.168.6.104:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:3311@192.168.6.100;ftag=925f0b001c0632395f2e8-34ee90e;lr =on>..Via: SIP/2.0/UDP 192.168.6.100;branch=0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;ta g=925f0b001c0632395f2e8-34ee90e..To: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..Call-ID: 000b5f92-63c00003-63cc121e- 248e7823@192.168.6.103..Date: Wed, 11 Feb 2004 18:33:53 GMT..CSeq: 101 ACK..Content-Length: 0.... ### U 128.163.55.134:496 -> 224.0.1.40:496 ..........7=........ ############ U 192.168.6.104:50020 -> 192.168.6.100:5060 BYE sip:3311@192.168.6.100:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.6.104:5060..From: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de -474d2b85..To: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..Call-ID: 000b5f92-63c00003-63cc121e-248e78 23@192.168.6.103..Date: Wed, 11 Feb 2004 18:34:02 GMT..User-Agent: Cisco-SIP-IP-Phone/2..CSeq: 101 BYE..Route: <sip:3322@192.168.6.103 :5060>..Content-Length: 0.... # U 192.168.6.100:5060 -> 192.168.6.103:5060 BYE sip:3322@192.168.6.103:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:3311@192.168.6.100;ftag=c25f0b001363563cc13de-474d2b85;l r=on>..Via: SIP/2.0/UDP 192.168.6.100;branch=z9hG4bK8153.ac3bbf61.0..Via: SIP/2.0/UDP 192.168.6.104:5060..From: <sip:3311@192.168.6.10 0>;tag=c25f0b001363563cc13de-474d2b85..To: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..Call-ID: 000b5 f92-63c00003-63cc121e-248e7823@192.168.6.103..Date: Wed, 11 Feb 2004 18:34:02 GMT..User-Agent: Cisco-SIP-IP-Phone/2..CSeq: 101 BYE..Co ntent-Length: 0.... # U 192.168.6.103:49834 -> 192.168.6.100:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.6.100;branch=z9hG4bK8153.ac3bbf61.0,SIP/2.0/UDP 192.168.6.104:5060..From: <sip:3311@192.168.6 .100>;tag=c25f0b001363563cc13de-474d2b85..To: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..Call-ID: 00 0b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Server: Cisco-SIP-IP-Phone/2..CSeq: 101 BYE..Content-Length: 0.... # U 192.168.6.100:5060 -> 192.168.6.104:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.6.104:5060..From: <sip:3311@192.168.6.100>;tag=c25f0b001363563cc13de-474d2b85..To: "User ID B alaji" <sip:3322@192.168.6.100>;tag=925f0b001c0632395f2e8-34ee90e..Call-ID: 000b5f92-63c00003-63cc121e-248e7823@192.168.6.103..Server: Cisco-SIP-IP-Phone/2..CSeq: 101 BYE..Content-Length: 0.... ###### exit Thanks in advance, Balaji

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[Serusers] Request for SER.CFG file
by Morris, Scott 12 Feb '04

12 Feb '04

I am battling this RADIUS Authentication configuration for SER. Does anyone have a SER.CFG file they can send me of a working RADIUS setup configuration file Scott Morris Enterprise Network Engineer DOE - ORAU / ORISE 865-576-4672

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RE: [Serusers] Motorola VT1000v with Ser
by Paul.Wasik＠ipc.com 11 Feb '04

11 Feb '04

Vonage locks the VoIP configuration menus on the VT1000v units they sell. Does anyone know where to get these units other than from Vonage or how to unlock the VoIP config menus? -----Original Message----- From: Alan Crosswell [mailto:alan@columbia.edu] Sent: Wednesday, February 11, 2004 5:47 PM To: Andy Singh Cc: 'serusers(a)lists.iptel.org' Subject: Re: [Serusers] Motorola VT1000v with Ser I guess Vonage switched from the Cisco ATA-186. Andy Singh wrote: > Just wondering if any tried using Motorola VT1000v box with Ser. It's a SIP > UA and you can plug in your analog phone into it to make calls. You can find > out more on this at > > http://broadband.motorola.com/consumers/products/vt1000v/ and company > called vonage sells the VOIP service using this to consumer. > > Thanks > Andy > > _______________________________________________ > 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 Important Notice ************************************************* This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail. E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Serusers] Re: Sip<>H323 proxy
by Atle Samuelsen 11 Feb '04

11 Feb '04

* Arnd Vehling <av(a)nethead.de> [040211 22:26]: > Youre sende address isnt valid. i..e camro instead of camaro Thanks.. Just one of my typos.. Ill talk my boss, and see if they are intrested in sponsoring the software.. Was'nt there a guy who said he where gonna do it a month or 2 ago ? or do I remeber wrong? - Atle > > -------- Original Message -------- > Subject: failure notice > Date: 11 Feb 2004 20:21:43 -0000 > From: MAILER-DAEMON(a)ns.voipnet.net > To: av(a)nethead.de > > Hi. This is the qmail-send program at ns.voipnet.net. > I'm afraid I wasn't able to deliver your message to the following addresses. > This is a permanent error; I've given up. Sorry it didn't work out. > > <clona(a)camro.no>: > Sorry, I couldn't find any host named camro.no. (#5.1.2) > > --- Below this line is a copy of the message. > > Atle Samuelsen wrote: > >He he.. Saw the NDA from sipquest.. Did'nt want to sign it myself.. > > Yes, quite annoying. Especially when considering that the product > is based on a former open source software (called siph323). > > Unluckily the siquest guys have been very successfull in removing > all open source copies from the net. At least i wasnt able to > find any. > > Right now i am talking to a guy who offered enhancing Vovidas > siph323csgw so it can bridge between SIP and H323 networks. > I am looking for other persons who are willing to financially > support this project. The costs would be between EU 500 and EU 300 > and the result would be returned to the open source community. > Details of this would need to be worked out though. > > Contact me if youre interested in this. > > -- Arnd > >

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[Serusers] Serving multiple domains
by Alan Litster 11 Feb '04

11 Feb '04

Hi List, We have a few local domains that we will be using SER act as a proxy and to protect our gateway. In the near future we will also be providing PSTN breakout for other SIP domains that are not on our proxy. What is the best way to allow traffic from certain know SIP domains out through our gateways? I've thought about having users authenticate to our proxy, but this could get very complex and messy serving multiple local & remote domains. Other option is just to allow traffic through from the trusted domains I guess. One last question, is any one familiar with the Cisco AS5300? How do you go about locking it down so that only authorised users can pass calls out onto the PSTN? We have a number of Vega 100's that have the option of only allow proxy invited calls. Is there something similar on the AS5300 or would that have to do authentication aswell, AAA ? I know that this question is slightly off topic, but with it being related I thought someone my be able to answer my question. Many Thanks, Alan ------------------------------------------------------------------------------------------------------- This email, and any files transmitted with it, is copyright and may contain confidential information. The contents are intended for the use of the addressee(s) only. Unauthorized use may be unlawful. If you receive this email by mistake, please advise sender immediately. The views of the author may not necessarily constitute the views of Telco Electronics Limited. Nothing in this mail shall bind Telco Electronics Limited in any contract or obligation. Telco Electronics Limited 6-8 Oxford Court Brackley Northants NN13 7XY Tel 07000 701999 Fax 07000 701777

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sr-users February 2004