sr-users October 2005

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214 participants
457 discussions

RE: [Serusers] Radius authentication and accounting issues

by deepak.chandrasekaran＠wipro.com

Hi harry, I don't find any error messages. But whenever I try to start SER, it crashes immediately. I am attaching the ser.cfg file herewith. Here is the log message: Nov 16 11:01:01 crond(pam_unix)[29639]: session closed for user root Nov 16 12:01:01 crond(pam_unix)[30131]: session opened for user root by (uid=0) Nov 16 12:01:01 crond(pam_unix)[30131]: session closed for user root Nov 16 13:01:01 crond(pam_unix)[30623]: session opened for user root by (uid=0) Nov 16 13:01:02 crond(pam_unix)[30623]: session closed for user root Nov 16 14:01:01 crond(pam_unix)[31115]: session opened for user root by (uid=0) Nov 16 14:01:02 crond(pam_unix)[31115]: session closed for user root Nov 16 14:14:23 ser: WARNING: fix_socket_list: could not rev. resolve 10.200.220.8 Nov 16 14:14:23 ser: WARNING: fix_socket_list: could not rev. resolve 10.200.220.8 Nov 16 14:14:23 ./ser[31240]: Maxfwd module- initializing Am I missing out something? Thanks & Regards, Deepak -----Original Message----- From: harry gaillac [mailto:gaillacharry@yahoo.fr] Sent: Monday, October 17, 2005 1:46 PM To: Deepak Chandrasekaran (WT01 - Voice & Next Generation Networks) Subject: RE: [Serusers] Radius authentication and accounting issues hello, Post you error message Harry --- deepak.chandrasekaran(a)wipro.com a écrit : > > Hi all, > > > > I am trying to provide radius authentication and > accounting in ser > 0.9.3. > > I am using freeradius 1.0.4. > > > > I have compiled SER successfully and got all .so > files. > > > > Every time I try to start ser, it crashes. > > Can anybody suggest some possible reasons as to why > this happens? > > > > I followed radius how-to doc to configure radius > server and > radiusclient-ng 0.5.1 library. > > > > Can anyone provide with sample ser.cfg file modified > to use radius > authentication and accounting? > > > > Any help would be greatly appreciated. > > > > Thanks in advance. > > > > Regards, > > Deepak > > > > > > > > Confidentiality Notice > > The information contained in this electronic message > and any attachments to this message are intended > for the exclusive use of the addressee(s) and may > contain confidential or privileged information. If > you are not the intended recipient, please notify > the sender at Wipro or Mailadmin(a)wipro.com > immediately > and destroy all copies of this message and any attachments.> _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

19 years, 1 month

[Serusers] only receive trying,but no ring

by fishman7788

hi I now work to make a call,receive the tring packet but now ring later,the packet is below,pls help. > Received from udp:10.110.11.9:5060 at 17/10/2005 16:46:47:875 (555 bytes): SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.110.10.149:1357;branch=z9hG4bK-rkad5409p17a;rport=1357 From: "test1" <sip:1234@10.110.11.9>;tag=b820xr0tze To: <sip:5678@10.110.11.9;user=phone> Call-ID: f56453439fd5-qaod8sww6dpi@snomSoft CSeq: 1 INVITE Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 10.110.11.9:5060 "Noisy feedback tells: pid=24182 req_src_ip=10.110.10.149 req_src_port=1357 in_uri=sip:5678@10.110.11.9;user=phone out_uri=sip:5678@0.0.0.0:1511;line=3tjczupg via_cnt==1" > 　　　　　　　　fishman7788 　　　　　　　　fishman7788(a)163.com 　　　　　　　　　　2005-10-17

19 years, 1 month

Re: [Serusers] serweb: DB Error: no such field

by Karel Kozlik

Hi, > Oct 14 18:23:12 serweb [error] file: > /usr/local/serweb/modules/subscribers/method.get_users.php:119: DB > Error: no such field - select s.username from subscriber s where > s.domain = 'vpbx.elmit.com' and true [nativecode=1054 ** Unknown column > 'true' in 'where clause'] Interesting. Which version of mysql are you useing? It seems some older version doesn't know the keyword 'true'. Thanks for report, I will fix it. >> For get latest stable version use this command: >> cvs -d:pserver:anonymous@cvs.berlios.de:/cvsroot/serweb co -P -r >> rel_0_9_2 serweb >> > I will try that, if we cannot get that to work ;-) I guess it is easier way for you :-) > Does 0_9_2 work with cvs of openser? Sorry, I don't know. May be yes, but I never tested it and I don't know details about openser. > I see on you email address iptel.org I read somewhere that there is a > full ser.cfg of iptel in the /etc directory for reference, but I did not > find it. I would like to study it. Although for the moment I just need: > Ser on port 5060. If enum, handle it, if not enum forward all to > asterisk on port 5061. It is in dir etc/obsoleted. But I'm not sure if these files are in released version of ser. If they isn't look at cvs: http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/obsoleted/ Karel

19 years, 1 month

[Users] Radius accounting and freeside

by harry gaillac

Hello, I try to setup ser-0.9.4 for accounting with freeradius-1.0.2 and mysql-4. Here is a part of my ser.cfg file # -- acc params -- modparam("acc", "db_url", "mysql://xxx:xxx@nxs.yi.org/ser") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_fmt", "miocfsp") modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "service_type", 16) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 1) modparam("acc", "radius_flag", 4) modparam("acc", "radius_missed_flag", 5) I set "setflag(4);" for INVITE BYE and CANCEL messages. Look at my sql.php.html file (no script) I can't fix Acct-Start-Time and Acct-Stop-Time to calculate Acct-Session-Time with freeside How can I do it ? Thanks for help Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 1 month

[Serusers] Radius authentication and accounting issues

by deepak.chandrasekaran＠wipro.com

Hi all, I am trying to provide radius authentication and accounting in ser 0.9.3. I am using freeradius 1.0.4. I have compiled SER successfully and got all .so files. Every time I try to start ser, it crashes. Can anybody suggest some possible reasons as to why this happens? I followed radius how-to doc to configure radius server and radiusclient-ng 0.5.1 library. Can anyone provide with sample ser.cfg file modified to use radius authentication and accounting? Any help would be greatly appreciated. Thanks in advance. Regards, Deepak Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

19 years, 1 month

[Serusers] Username conventions with Asterisk as voicemail ???

by Daryl Sanders

I'm looking for pointers, best practices or experiences from SER users who are using Asterisk for voicemail-only in their SER setup, especially when it comes to planning username/extension/mailbox naming schemes. This is for a large-scale deployment (10K+ users) so I want to get it right the first time. Many of my users will have DIDs tied to their account while others will not. Are there any advantages/disadvantages to having alpha-numeric SER usernames and using aliases for DID numbers? I'm not sure if it matters, but we will be using a third party PSTN provider and will need to support DIDs from multiple countries. Also, are there any scripts available to automatically create an Asterisk voicemail box when a new user is added to SER? As always, your input is greatly appreciated! - Daryl

19 years, 1 month

[Serusers] Changeing a number dialed for pstn

by Mr Greg Plater

Hi Does anyone know how I can make a change in my ser.cfg file to change a number dialed Example: Someone dials 738211234 And I wish to send the number out to pstn as 61738211234 I use the following to send out 61738211234 but I would rather shorten the numbers dialed by a user and have the 61 auto added. if (uri=~"^sip:[0-9]{11}@") { # Domestic PSTN Australia wide route(4); break; }; # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("202.173.179.228"); # PSTN GATEWAY IP ADDRESS Asterisk Bridge avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] {

19 years, 1 month

[Serusers] Prepaid Service

by Ryan Pagquil

Hi Guys, I'm wondering how can I terminate active SIP sessions when they already consumed their credits. Because I want to make a prepaid service for VoIP. Is there any known application that can implement prepaid service for VoIP? Please help on this... Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

19 years, 1 month

[Serusers] SIPSAK

by Rick Thompson

Just started to try SIPSAK and I'm getting the following error Sipsak -T -s sip:1079#a.b.c.d:5060 warning: IP extract from warning activated to be more informational error: this FQDN or IP is not valid: rtlaptop does anyone know why? Thanks Richard C. Thompson

19 years, 1 month

[Serusers] TCP/TLS and NAT. Problem with BYE...

by Alexander Ph. Lintenhofer

Hi everybody, A short question: alice(a)atlanta.com is behind NAT. She uses TCP to connect her UAC to sip.atlanta.com in the public network. Inviting bob(a)biloxi.com suceeds. But Bob can't terminate the call. His BYE is not routed back, because fix_nated_contact() does only work with UDP. And Bob builds the R-URI with the Infos of Alice's contact header. The route is as follows: alice(a)atlanta.com [172.16.0.4] -> natbox.atlanta.com [172.16.0.1 | 192.168.0.13] -> sip.atlanta.com [192.168.0.14] -> sip.biloxi.com [192.168.1.14] -> bob(a)biloxi.com [192.168.1.1] Possible solutions: - force_tcp_alias() -> but reading the draft draft-ietf-sip-connect-reuse-04.txt yields that this must be supported by the components - so doesn't the snom360 of Alice - commenting some lines in nathelper.c and activate it for TCP -> this works pretty fine and the BYE finds its way from Bob through the NAT-Box to Alice. But this is a dirty solution. So does anybody has a similar problem? My config works fine with UDP but switching to TCP makes life hard... I attached the traces Bob received, his last BYE is finally dropped by sip.atlanta.com, because the network-address 172.16.0.4 can't be found! Thank you for your help! regards, Philipp ========================================================================= Received from tcp:192.168.1.14:5060 at 15/10/2005 19:18:32:250 (1751 bytes): INVITE sip:bob@192.168.1.1:2063;transport=tcp;line=wxqurd1s SIP/2.0 Record-Route: <sip:192.168.1.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.1.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Via: SIP/2.0/TCP 192.168.1.14;branch=z9hG4bKcad9.057cd815.0;i=d Via: SIP/2.0/TLS 192.168.0.14:5061;branch=z9hG4bKcad9.685be3c3.0;i=1 Via: SIP/2.0/TCP 172.16.0.4:2327;received=192.168.0.13;branch=z9hG4bK-y79imu6dlqxs;rport=2327 From: "Alice" <sip:alice@atlanta.com>;tag=5s8qncdbso To: <sip:bob@biloxi.com> Call-ID: 3c2675cac832-ce5ge5sxlx2q@snom360 CSeq: 1 INVITE Max-Forwards: 68 Contact: <sip:alice@172.16.0.4:2327;transport=tcp;line=fyyuh6tl> P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/4.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600 Content-Type: application/sdp Content-Length: 507 P-hint: outbound P-hint: forced TLS relay P-hint: usrloc applied Sent to tcp:192.168.1.14:5060 at 15/10/2005 19:18:32:270 (929 bytes): SIP/2.0 180 Ringing Via: SIP/2.0/TCP 192.168.1.14;branch=z9hG4bKcad9.057cd815.0;i=d Via: SIP/2.0/TLS 192.168.0.14:5061;branch=z9hG4bKcad9.685be3c3.0;i=1 Via: SIP/2.0/TCP 172.16.0.4:2327;received=192.168.0.13;branch=z9hG4bK-y79imu6dlqxs;rport=2327 Record-Route: <sip:192.168.1.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.1.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> From: "Alice" <sip:alice@atlanta.com>;tag=5s8qncdbso To: <sip:bob@biloxi.com>;tag=fxdufnc4xz Call-ID: 3c2675cac832-ce5ge5sxlx2q@snom360 CSeq: 1 INVITE Contact: <sip:bob@192.168.1.1:2063;transport=tcp;line=wxqurd1s> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 Sent to tcp:192.168.1.14:5060 at 15/10/2005 19:18:33:390 (1377 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.1.14;branch=z9hG4bKcad9.057cd815.0;i=d Via: SIP/2.0/TLS 192.168.0.14:5061;branch=z9hG4bKcad9.685be3c3.0;i=1 Via: SIP/2.0/TCP 172.16.0.4:2327;received=192.168.0.13;branch=z9hG4bK-y79imu6dlqxs;rport=2327 Record-Route: <sip:192.168.1.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.1.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> From: "Alice" <sip:alice@atlanta.com>;tag=5s8qncdbso To: <sip:bob@biloxi.com>;tag=fxdufnc4xz Call-ID: 3c2675cac832-ce5ge5sxlx2q@snom360 CSeq: 1 INVITE Contact: <sip:bob@192.168.1.1:2063;transport=tcp;line=wxqurd1s> Require: timer Session-Expires: 3600;refresher=uac User-Agent: snom360/4.3 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 296 ACK sip:bob@192.168.1.1:2063;transport=tcp;line=wxqurd1s SIP/2.0 Record-Route: <sip:192.168.1.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.1.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Record-Route: <sip:192.168.0.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Via: SIP/2.0/TCP 192.168.1.14;branch=0;i=d Via: SIP/2.0/TLS 192.168.0.14:5061;branch=0;i=1 Via: SIP/2.0/TCP 172.16.0.4:2327;received=192.168.0.13;branch=z9hG4bK-o9f1lglhf4pk;rport=2327 From: "Alice" <sip:alice@atlanta.com>;tag=5s8qncdbso To: <sip:bob@biloxi.com>;tag=fxdufnc4xz Call-ID: 3c2675cac832-ce5ge5sxlx2q@snom360 CSeq: 1 ACK Max-Forwards: 68 Contact: <sip:alice@172.16.0.4:2327;transport=tcp;line=fyyuh6tl> Content-Length: 0 P-hint: rr-enforced P-hint: rr-enforced Sent to tcp:192.168.1.14:5060 at 15/10/2005 19:18:34:480 (703 bytes): BYE sip:alice@172.16.0.4:2327;transport=tcp;line=fyyuh6tl SIP/2.0 Via: SIP/2.0/TCP 192.168.1.1:2063;branch=z9hG4bK-gu03sll9uumm;rport Route: <sip:192.168.1.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> Route: <sip:192.168.1.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Route: <sip:192.168.0.14:5061;transport=tls;r2=on;ftag=5s8qncdbso;lr=on> Route: <sip:192.168.0.14;transport=tcp;r2=on;ftag=5s8qncdbso;lr=on> From: <sip:bob@biloxi.com>;tag=fxdufnc4xz To: "Alice" <sip:alice@atlanta.com>;tag=5s8qncdbso Call-ID: 3c2675cac832-ce5ge5sxlx2q@snom360 CSeq: 1 BYE Max-Forwards: 70 Contact: <sip:bob@192.168.1.1:2063;transport=tcp;line=wxqurd1s> User-Agent: snom360/4.3 Content-Length: 0

19 years, 1 month

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sr-users October 2005