sr-users December 2003

sr-users@lists.kamailio.org

108 participants
257 discussions

[Serusers] digit strip problem
by Bo 19 Dec '03

19 Dec '03

Hi All, I added the following two lines in ser.cfg. strip(0); prefix("99"); The purpose is to strip the leading '0' first then add the prefix '99'. But it just added '99' and didn't strip the '0'. Does the command 'strip' still work? Thanks, Bob

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[Serusers] 483
by jerk face 18 Dec '03

18 Dec '03

Here's the scenario: As of now, I can only get users to register with my SER if I don't use database authentication. These users must also have a static IP address to register. I am now trying to get a client to register with SER from behind a NAT. I am using a STUN server address to help with the NAT situation. When I do this I get a 483 message. >From looking at similar posts, the problem is usually caused by setting uri=="some.domain.com" I don't have this configuration; in ser.cfg I have the following: alias="my.domain.com" if(uri==myself) Also, I am running ser-0.8.12 if that makes any difference. Does anybody know why I am getting 483 error messages? Any help is always appreciated. __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

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RE: [Serusers] request
by Subramaniam, Girija 18 Dec '03

18 Dec '03

The server seems to be slow. It took me a few days of repeated trials to successfully download the server. -girija -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Thursday, December 18, 2003 5:10 PM To: Meril Vasantha Fernandopulle Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] request http://iptel.org/ser/download/ Jan. On 18-12 00:17, Meril Vasantha Fernandopulle wrote: > dear sir > > I am interesting about SIP Express Router and I try to > download it several times but I could not download it. > So please advice me a good way to do it. > or send me by mail. > looking forward your reply. > > regards > > vasantha > > __________________________________ > Do you Yahoo!? > New Yahoo! Photos - easier uploading and sharing. > http://photos.yahoo.com/ > > _______________________________________________ > 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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Re: [Serusers] Using asterisk as a voicemail server with SER
by Samy Touati 18 Dec '03

18 Dec '03

Hi, I saw messages complaining about no email addresses (even if there was one). I saw some postings about this topic and multi-domain, but I only have one domain, beside this message appear after a while. I'm going to upgrade to 0.8.12 and redo more tests. But the main issue in my opinion is the lack of IVR to retrieve the messages without using a computer. I'm trying to see if asterisk can support acting as a media server only. I want to have ser the sip proxy server where all the phones register with and asterisk being only the voicemail where calls get forwarded to from ser when there is failure or the user is busy. I don't know if it's feasable, according to what I see so far it doesn't look like it support this kind of config. Can you confirm this? So I'm left with not much option here if I want to have IVR support for voicemail: * Try to implement it in sems, what is your estimate on the complexity of this task, do you have any suggestions on tacking this task ? Thanks. Samy. ----- Original Message ----- From: Jan Janak <jan(a)iptel.org> Date: Thursday, December 18, 2003 3:31 pm Subject: Re: [Serusers] Using asterisk as a voicemail server with SER > On 16-12 09:31, Samy Touati wrote: > > Hi, > > > > I installed ser 0.8.11.rc1 along with the sems tied to the same > version.> Everything installed fine and was running on the same > host with different > > port numbers. > > The instability comes from the fact that the ser configured to > receive voice > > calls will just stop answering them after some time. > > I looked through ethereal and no abnormal messages were generated: > > I see the invite going to ser-vm on port 5071, if the user is > busy or not > > answering, I see the trying/ringing and the caller hears the > ringing. After > > a timeout the ringing stops and I get a fast busy with a 408. > > Now restarting ser and ser-vm puts everything back on track, but > after a > > couple of hours I see this behavior again. > > Sounds strange. Is there any message in the logs of ser ? Ser should > log some error messages if it is not working. Could you check > that ? > > Jan. >

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[Serusers] Unformatted Exerpts from for BYE problem
by Darren Sessions 18 Dec '03

18 Dec '03

SIP/2.0 200 OK To: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642 From: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3 Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157 CSeq: 101 INVITE Via: SIP/2.0/UDP 63.86.212.154;branch=z9hG4bKb2f4.035.0 Via: SIP/2.0/UDP 63.86.212.157:5060 Record-Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr> Timestamp: 1071766157 Contact: John Walter <sip:8646783182@63.86.212.155:5060> Server: Sipura/SPA2000-1.0.20 Content-Length: 150 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel Content-Type: application/sdp v=0 o=- 88599 88599 IN IP4 192.168.0.101 s=- c=IN IP4 63.86.212.155 t=0 0 m=audio 16404 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ?Úá?ÉFóóªûì ^NEå"8@þÁ?VÔsÀ¨eÄÄÑBEACK sip:8646783182@63.86.212.155:5060 SIP/2.0 Record-Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr> Via: SIP/2.0/UDP 63.86.212.154;branch=0 Via: SIP/2.0/UDP 63.86.212.157:5060 From: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3 To: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642 Date: Thu, 18 Dec 2003 16:49:17 GMT Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157 Max-Forwards: 5 Content-Length: 0 CSeq: 101 ACK À¨e?VÔsÄÄ·á}BYE sip:8644679887@63.86.212.157:5060 SIP/2.0 Via: SIP/2.0/UDP 63.86.212.155:5060;branch=z9hG4bK-b76fe29f From: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642 To: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3 Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157 CSeq: 101 BYE Max-Forwards: 70 Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr> User-Agent: Sipura/SPA2000-1.0.20 Content-Length: 0 Úá?ªûì ^NE":@þ%?VÔsÀ¨eÄÄkè¯SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 63.86.212.155:5060;branch=z9hG4bK-b76fe29f From: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642 To: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3 Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157 CSeq: 101 BYE Contact: <sip:8644679887@63.86.212.148> Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 63.86.212.148:5060 "Noisy feedback tells: pid=20033 req_src_ip=63.86.212.154 req_src_port=5060 in_uri=sip:8644679887@63.86.212.148:5060 out_uri=sip:8644679887@63.86.212.148:5060 via_cnt==0" -----Original Message----- From: Andres [ mailto:andres@telesip.net <mailto:andres@telesip.net> ] Sent: Thursday, December 18, 2003 3:25 PM To: Darren Sessions; serusers(a)lists.iptel.org Subject: Re: [Serusers] BYE Message Problem On Thursday 18 December 2003 15:11, Darren Sessions wrote: > Sipura Firmware is 1.0.20 > > We are already in talks with Sipura about this issue as we found a bug for > them yesterday. It appears though based on the Etheral traces and ngrep > that the Sipura is sending the BYE messages - but SER is not forwarding > them along. Can you post all the messages here(INVITE...to ...BYE)? Otherwise it is hard to guess where the problem might be. I'm sure Jiri or Jan can spot the error right away. > > We are using Ser 8.12 on a Sun Ultra 60 - and as I said before, all other > instances work flawlessly. > > Thanks, > > - Darren > > -----Original Message----- > From: Andres [ mailto:andres@telesip.net <mailto:andres@telesip.net> ] > Sent: Thursday, December 18, 2003 2:58 PM > To: Darren Sessions; serusers(a)lists.iptel.org > Subject: Re: [Serusers] BYE Message Problem > > On Thursday 18 December 2003 14:50, Darren Sessions wrote: > > We also have a problem with BYE's not being sent from an on-net call > > (sipura to sipura). > > What firmware version are you using? We have been testing the SPA2000 for > about 3 weeks now and have not seen this issue. > > > -----Original Message----- > > From: Darren Sessions > > Sent: Thursday, December 18, 2003 2:32 PM > > To: 'serusers(a)lists.iptel.org' > > Subject: RE: Serusers Digest, Vol 8, Issue 16 > > Importance: High > > > > > > Situation: > > > > Endpoint is called Party (Sipura SPA2000) > > > > Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640 > > router then sent to SER. > > > > Endpoint receives call - everything works. When the Sipura hangs up the > > call, it sends a BYE to ser - but ser does not send the BYE to the router > > to disconnect the TDM channel. > > > > All other situations result in a completed call with channels released on > > the router just fine. > > > > After further inspection - we noticed that when the Sipura hangs up and > > send the BYE to SER, SER responds with this: > > > > > > SIP/2.0 481 Call/Transaction Does Not Exist > > Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f > > From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642 > > To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3 > > Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157 > > CSeq: 101 BYE > > Contact: <sip:8644679887@44.44.444.148> > > Server: Sip EXpress router (0.8.12 (i386/linux)) > > Content-Length: 0 > > Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033 > > req_src_ip=22.22.222.154 req_src_port=5060 > > in_uri=sip:8644679887@44.44.444.148:5060 > > out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0" > > > > > > Any ideas? > > > > Thanks, > > > > - Darren > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers>

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[Serusers] Debug information
by Rick Gocher 18 Dec '03

18 Dec '03

Hi All, Just me, still trying to get authentication working... I have found by placing a high debug value and starting ser I received some errors I never saw before as I had no ata box plugged in during the startup. I'm not sure if this points to why I can't authenticate. When the output says there is no routing headers, is that the fault of my ATA? Dec 11 16:07:19 sip ser: 8(4583) SIP Request: Dec 11 16:07:19 sip ser: 8(4583) method: <REGISTER> Dec 11 16:07:19 sip ser: 8(4583) uri: <sip:sip.coptalk.com> Dec 11 16:07:19 sip ser: 8(4583) version: <SIP/2.0> Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=1 Dec 11 16:07:19 sip ser: 8(4583) end of header reached, state=5 Dec 11 16:07:19 sip ser: 8(4583) parse_headers: Via found, flags=1 Dec 11 16:07:19 sip ser: 8(4583) parse_headers: this is the first via Dec 11 16:07:19 sip ser: 8(4583) After parse_msg... Dec 11 16:07:19 sip ser: 8(4583) preparing to run routing scripts... Dec 11 16:07:19 sip ser: 8(4583) entering main route 8(4583) DEBUG : is_maxfwd_present: searching for max_forwards header Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=128 Dec 11 16:07:19 sip ser: 8(4583) end of header reached, state=8 Dec 11 16:07:19 sip ser: 8(4583) DEBUG: get_hdr_field: <To> [27]; uri=[sip:Rick@sip.coptalk.com] Dec 11 16:07:19 sip ser: Dec 11 16:07:19 sip ser: ] Dec 11 16:07:19 sip ser: 8(4583) get_hdr_field: cseq <CSeq>: <7> <REGISTER> Dec 11 16:07:19 sip ser: 8(4583) DEBUG: get_hdr_body : content_length=0 Dec 11 16:07:19 sip ser: 8(4583) found end of header Dec 11 16:07:19 sip ser: 8(4583) DEBUG: is_maxfwd_present: max_forwards header not found! Dec 11 16:07:19 sip ser: 8(4583) DEBUG: add_param: tag=1804086518 Dec 11 16:07:19 sip ser: 8(4583) end of header reached, state=29 Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=256 Dec 11 16:07:19 sip ser: 8(4583) find_first_route(): No Route headers found Dec 11 16:07:19 sip ser: 8(4583) loose_route(): There is no Route HF Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if host==us: 16==9 && [sip.coptalk.com] == [127.0.0.1] Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if port 5060 matches port 5060 Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if host==us: 16==13 && [sip.coptalk.com] == [64.189.165.205] Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if port 5060 matches port 5060 Dec 11 16:07:19 sip ser: 8(4583) request for registration 8(4583) check_nonce(): comparing [3fd907c2aa2d6c72cd666f9109e45155055b2d36] and [3fd907c23a57f75133a55828041f2c9ee7b081c2] Dec 11 16:07:19 sip ser: 8(4583) pre_auth(): Invalid nonce value received, very suspicious ! Dec 11 16:07:19 sip ser: Dec 11 16:07:19 sip ser: ' Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=-1 Dec 11 16:07:19 sip ser: 8(4583) check_via_address(64.189.165.206, 64.189.165.206, 0) Dec 11 16:07:19 sip ser: 8(4583) receive_msg: cleaning up Dec 11 16:07:19 sip ser: 6(4581) SIP Request: Dec 11 16:07:19 sip ser: 6(4581) method: <REGISTER> Dec 11 16:07:19 sip ser: 6(4581) uri: <sip:sip.coptalk.com> Dec 11 16:07:19 sip ser: 6(4581) version: <SIP/2.0> Dec 11 16:07:19 sip ser: 6(4581) parse_headers: flags=1 Dec 11 16:07:19 sip ser: 6(4581) end of header reached, state=5 Dec 11 16:07:19 sip ser: 6(4581) parse_headers: Via found, flags=1 Dec 11 16:07:19 sip ser: 6(4581) parse_headers: this is the first via Dec 11 16:07:19 sip ser: 6(4581) After parse_msg... Dec 11 16:07:19 sip ser: 6(4581) preparing to run routing scripts... Dec 11 16:07:19 sip ser: 6(4581) entering main route 6(4581) DEBUG : is_maxfwd_present: searching for max_forwards header Dec 11 16:07:19 sip ser: 6(4581) parse_headers: flags=128 Dec 11 16:07:19 sip ser: 6(4581) end of header reached, state=8 Dec 11 16:07:19 sip ser: 6(4581) DEBUG: get_hdr_field: <To> [27]; uri=[sip:Rick@sip.coptalk.com] Dec 11 16:07:19 sip ser: Dec 11 16:07:19 sip ser: ] Dec 11 16:07:19 sip ser: 6(4581) get_hdr_field: cseq <CSeq>: <8> <REGISTER> Dec 11 16:07:19 sip ser: 6(4581) DEBUG: get_hdr_body : content_length=0 Dec 11 16:07:19 sip ser: 6(4581) found end of header Dec 11 16:07:19 sip ser: 6(4581) DEBUG: is_maxfwd_present: max_forwards header not found! Dec 11 16:07:19 sip ser: 6(4581) DEBUG: add_param: tag=1804086518 Dec 11 16:07:19 sip ser: 6(4581) end of header reached, state=29 Dec 11 16:07:19 sip ser: 6(4581) parse_headers: flags=256 Dec 11 16:07:19 sip ser: 6(4581) find_first_route(): No Route headers found Dec 11 16:07:19 sip ser: 6(4581) loose_route(): There is no Route HF Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if host==us: 16==9 && [sip.coptalk.com] == [127.0.0.1] Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if port 5060 matches port 5060 Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if host==us: 16==13 && [sip.coptalk.com] == [64.189.165.205] Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if port 5060 matches port 5060 Dec 11 16:07:20 sip ser: 6(4581) request for registration 6(4581) check_nonce(): comparing [3fd907e3929aea9bad25fbd47b3ea062d9215574] and [3fd907e3929aea9bad25fbd47b3ea062d9215574] Dec 11 16:07:20 sip ser: 6(4581) get_ha1(): no result for user 'Rick(a)coptalk.com' Dec 11 16:07:20 sip ser: Dec 11 16:07:20 sip ser: ' Dec 11 16:07:20 sip ser: 6(4581) parse_headers: flags=-1 Dec 11 16:07:20 sip ser: 6(4581) check_via_address(64.189.165.206, 64.189.165.206, 0) Dec 11 16:07:20 sip ser: 6(4581) receive_msg: cleaning up

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[Serusers] Re: SER AND ASTERISK
by listas iPfone 18 Dec '03

18 Dec '03

HI all thanks for all replyes on my question about ser and asterisk.. Now i have that problem when i try to register a user in serweb: Warning: fopen("/tmp/ser_fifo", "w") - Permission denied in /var/www/html/functions.php on line 206 sipserver.com.br Gerenciamento de usuários sorry -- cannot open write fifo We regret but your sipserver.com.br confirmation attempt failed. Please contact info(a)ipfone.com.br for further assistance. What can be the problem? regards Miklos ----- Original Message ----- From: listas iPfone To: serusers(a)lists.iptel.org Sent: Wednesday, December 17, 2003 6:01 PM Subject: SER AND ASTERISK Hi All! I´m trying to run ser as my sip server and asterisk as my pbx in the same machine with acess to pstn. My goal is to have external users registering in ser ( with my dyndns - sipserver.com.br) and redirect the calls to my estensions in asterisk. How to do it? When i run ser, asterisk don´t find my phones anymore. I have two nic in this machine 192.168.0.31/37: [root@localhost root]# ser Listening on 127.0.0.1 [127.0.0.1]:5060 192.168.0.37 [192.168.0.37]:5060 192.168.0.31 [192.168.0.31]:5060 Aliases: localhost:5060 localhost.localdomain:5060 i think that i need to make ser listen to one ip and asterisk to other. If that is rigth, how can i make ser listen only on 192.168.0.31:5060? Thanks for all Miklos

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[Serusers] request
by Meril Vasantha Fernandopulle 18 Dec '03

18 Dec '03

dear sir I am interesting about SIP Express Router and I try to download it several times but I could not download it. So please advice me a good way to do it. or send me by mail. looking forward your reply. regards vasantha __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

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[Serusers] 401 Troubles
by jerk face 18 Dec '03

18 Dec '03

##I did try to post this question before, but it didn't get through.## Yet another question. I'm trying to get Windows Messenger to register with my ser, but I can't get it to work. Here are my configurations: export SIP_DOMAIN="mydomain.com" [Selected portions of ser.cfg] alias="demoSIP.mydomain.com" #---module loading--- if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("demoSIP.mydomain.com", "subscriber")) { www_challenge("demoSIP.mydomain.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; When I run ngrep -n 5060 -d eth0 myAccount I get the following: U myMessengerIPaddress:1082 -> mySERipAddress:5060 REGISTER sip:demoSIP.mydomain.com SIP/2.0..Via: SIP/2.0/UDP myMessengerIPaddress:6946..From: <sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189- 0a01-40b6-8b60-cdb0163da022..To: <sip:myAccount@demoSIP.mydomain.com>..Call-ID: 50fab216-4377-4432-962d-e324ad46b035@myMessengerIPaddress..CSeq : 1 REGISTER..Contact: <sip:myMessengerIPaddress:6946>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"..User-Agen t: Windows RTC/1.0..Expires: 1200..Event: registration..Allow-Events: presence..Content-Length: 0.... # U mySERipAddress:5060 -> myMessengerIPaddress:6946 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP myMessengerIPaddress:6946..From: <sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189-0a01-40b6-8b60-cd b0163da022..To: <sip:myAccount@demoSIP.mydomain.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c1f1..Call-ID: 50fab216-4377-4432-962d-e324ad4 6b035@myMessengerIPaddress..CSeq: 1 REGISTER..WWW-Authenticate: Digest realm="demoSIP.mydomain.com", nonce="3fe0cac641974960b96106d8e436ce1c0a3 713c8"..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 mySERipAddress:5060 "Noisy feedback tells: pid =3922 req_src_ip=myMessengerIPaddress req_src_port=1082 in_uri=sip:demoSIP.mydomain.com out_uri=sip:demoSIP.mydomain.com via_cnt==1".... ### U myMessengerIPaddress:1082 -> mySERipAddress:5060 REGISTER sip:demoSIP.mydomain.com SIP/2.0..Via: SIP/2.0/UDP myMessengerIPaddress:6946..From: <sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189- 0a01-40b6-8b60-cdb0163da022..To: <sip:myAccount@demoSIP.mydomain.com>..Call-ID: 50fab216-4377-4432-962d-e324ad46b035@myMessengerIPaddress..CSeq : 2 REGISTER..Contact: <sip:myMessengerIPaddress:6946>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"..User-Agen t: Windows RTC/1.0..Expires: 1200..Event: registration..Allow-Events: presence..Authorization: Digest username="myAccount", realm="demoSIP.mydomain.com", algorithm="md5", uri="sip:demoSIP.mydomain.com", nonce="3fe0cac641974960b96106d8e436ce1c0a3713c8", response="a03d6b206 d71f54674b5a48497b3e8aa"..Content-Length: 0.... ### U mySERipAddress:5060 -> myMessengerIPaddress:6946 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP myMessengerIPaddress:6946..From: <sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189-0a01-40b6-8b60-cd b0163da022..To: <sip:myAccount@demoSIP.mydomain.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c1f1..Call-ID: 50fab216-4377-4432-962d-e324ad4 6b035@myMessengerIPaddress..CSeq: 2 REGISTER..WWW-Authenticate: Digest realm="demoSIP.mydomain.com", nonce="3fe0cac641974960b96106d8e436ce1c0a3 713c8"..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 mySERipAddress:5060 "Noisy feedback tells: pid =3932 req_src_ip=myMessengerIPaddress req_src_port=1082 in_uri=sip:demoSIP.mydomain.com out_uri=sip:demoSIP.mydomain.com via_cnt==1".... I followed the documentation in seruser.pdf regarding the messenger bug. Any help is appreciated. Thank you for your time. __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

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[Serusers] Newbie with 401-itis disease
by jerk face 18 Dec '03

18 Dec '03

Yet another question. I'm trying to get Windows Messenger to register with my ser, but I can't get it to work. Here are my configurations: export SIP_DOMAIN="mydomain.com" [Selected portions of ser.cfg] alias="demoSIP.mydomain.com" #---module loading--- if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("mydomain.com", "subscriber")) { www_challenge("mydomain.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; When I run ngrep -n 5060 -d eth0 myAccount I get the following: U WindowsMessengerIP:1061 -> mySERip:5060 REGISTER sip:demoSIP.mydomain.com SIP/2.0..Via: SIP/2.0/UDP mySERip:6946.. From: <sip:myAccount@demoSIP.mydomain.com>;tag=57bd73f9-1d75-4da2-a0d4 -d3ed0acb4da3..To: <sip:myAccount@demoSIP.mydomain.com>..Call-ID: ec1d143e- 26a9-4144-9b6f-58af38b522f7@67.70.231.221..CSeq: 1 REGISTER..Contact: <sip: 67.70.231.221:6946>;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 mySERip:5060 -> WindowsMessengerIP:6946 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP WindowsMessengerIP:6946.. From:<sip:myAccount@demoSIP.mydomain.com>;tag=57bd73f9-1d75-4da2-a0d4-d3ed0acb4da3 To: <sip:myAccount@demoSIP.mydomain.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58 .c1f1..Call-ID: ec1d143e-26a9-4144-9b6f-58af38b522f7@67.70.231.221..CSeq: 1 REGISTER..WWW-Authenticate: Digest realm="mydomain.com", nonce="3fe0b9858a bc49c954578b998d4b1f709a6160f6"..Server: Sip EXpress router (0.8.12 (i386/l inux))..Content-Length: 0..Warning: 392 mySERip:5060 "Noisy feedback tells: pid=3495 req_src_ip=67.70.231.221 req_src_port=1061 in_uri=sip:demoSIP.mydomain.com out_uri=sip:demoSIP.mydomain.com via_cnt==1".... Does anybody know why the communication stops after the first 401? Any help is appreciated. Thank you for your time. ===== Asterisk is my lover, and IAX2 is our scented lubricant __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

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sr-users December 2003