sr-users April 2005

sr-users@lists.kamailio.org

250 participants
697 discussions

by info＠beeplove.com

> > Make sure you are not behind a Symmetric NAT. If so, you're > > dead. STUN does not work with Symmetric NAT. > > If a UA is behind Symmetric NAT, and > UA use STUN, and > SER have [RTP/Media]Proxy to handle Symmetric NAT, > this UA should be fine, right? Yes, but, if UA is behind symmetric NAT, I would not configure STUN to it. I'd just led mediaproxy solve the problem. Regards, Lucas But if you have 100 clients, it would be hard to put all clients in one group. Mohammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

19 years, 5 months

[Serusers] Tags, forwarding, redirects, and picky UACs

by Alistair Cunningham

I'm trying to implement forwarding to voicemail with a UAC that's picky about the tags in the To: headers it receives. First, I tried the following: failure_route [1] { rewritehostport("192.168.1.1:5060"); append_branch(); t_relay(); } This failed because the UAC received a "180 Ringing" from the ringing Grandstream phone via ser that contained a To: tag. After the timeout, it received another 180 from Asterisk (the voicemail system) with a different tag. Since it doesn't support forked INVITES, it dropped the call. Then I tried: failure_route [1] { rewritehostport("192.168.1.1:5060"); t_reply( "302", "Moved Temporarily" ); } This time it receives the 180 from the phone as before. Then it receives the 302. Alas, SER has ignored the tag from the phone, and created a new one of it's own, so the UAC again drops the call. Does anyone know of a way round this? Can SER be persuaded to use a tag it has already received? Can I strip the tag from the To: header in one or both of the responses? I'm willing to get my hands dirty with the SER source code if necessary. -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/

19 years, 5 months

[Serusers] No response in serctl add with password heslo

by Che Sosa

I tried adding subscribers using serctl: serctl add che che123 che@localhost It prompts for a MySql password. When I use "heslo", I get NO response. I tried changing the password of admin using: serctl passwrd admin heslo123. This command works with password heslo. What possibly went wrong? These are the config vars in ser_mysql.sh: DBNAME=ser DBHOST=localhost USERNAME=ser DEFAULT_PW=heslo ROUSER=serro RO_PW=47serro11 SQL_USER="root" CMD="mysql -h $DBHOST -u$SQL_USER " DUMP_CMD="mysqldump -h $DBHOST -u$SQL_USER -c -t " BACKUP_CMD="mysqldump -h $DBHOST -u$SQL_USER -c " TABLE_TYPE="TYPE=MyISAM" system info: Linux 2.6.8.1-12mdk #1 Fri Oct 1 12:53:41 CEST 2004 i686 AMD Athlon(TM) XP 1800+ unknown GNU/Linux Using: ser-0.9.0 ser -V output: flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_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 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 18:39:55 Apr 6 2005 with gcc 3.4 attached: ser config file

19 years, 5 months

[Serusers] Still no Help -- No ringback for ISDN->SIP->ISDN" using Cisco 2600 Gateway

by karl

Dear all, Following my previous email below, I am still facing the same problem without success of any solution. In this respect, I would very much appreciate whether someone who has come across the same problem in the past and possibly solved it, could at least possible tell me if it was a Cisco IOS issue or a ser configuration issue. Many thanks in advanced. Regards, Karl karl <ser_newbie(a)yahoo.com> wrote: Date: Fri, 8 Apr 2005 03:52:06 -0700 (PDT) From: karl <ser_newbie(a)yahoo.com> To: serusers(a)lists.iptel.org Subject: [Serusers] Re [cisco-voip] No early media for ISDN->SIP->ISDN Hello all, I have just come across the below email by Andreas Graning re the above subject, and its just exactly the same problem I am currently faced with. In this respect, I was wondering if anyone has in the meantime come across a solution to the issue. Any help on this subject shall be greatly appreciated. Thankssssssssssss. Regards, Karl [cisco-voip] No early media for ISDN->SIP->ISDNAndreas Granig SIP->ISDN" href="mailto:cisco-voip%40puck.nether.net?Subject=%5Bcisco-voip%5D%20No%20early%20media%20for%20ISDN-%3ESIP-%3EISDN&In-Reply-To=">a.granig at inode.at Wed May 12 09:52:19 EDT 2004 Previous message: [cisco-voip] CCM 3.2 and call Parks.. Next message: [cisco-voip] Domain name Change Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] --------------------------------- Hi,I've a Cisco 5300 running firmware image c5300-js-mz.122-15.T12.bin that interacts with SipExpressRouter (SER) from iptel.org.Following scenario: a caller from ISDN calls a number which is routed thru the C5300 to SER, where it is forwarded back to ISDN via the C5300 again. In this, and only in this scenario, the caller doesn't hear any early media (no ringback, no announcements etc).It works like charm when I for example call from a Cisco ATA to ISDN via C5300 and vice versa.When I configure "voice call send-alert" at the C5300, the PROGRESS is converted to ALERT and I hear a ringback tone, but other early media like announcements are also overwritten by a ringback tone.I've also tried "progress_ind setup enable 3", 8 for alert and 8 for proceed, without any success.I've alread studied (hopefully) all available Cisco documentation (voice commands for Firmware 12.2-T, the "Interworking Signaling Enhancements for H.323 and SIP VoIP", the "PSTN Callers not Hearing any Ring Back When they Call IP Phones", "Troubleshooting No Ringback Tone on ISDN-VoIP (H.323) Calls" and so on).Any hints?Regards,AndyPS: I've attached the Q931-debug and here are also the relevant parts of the C5300 configuration:voice call send-alert!voice-port 0:D input gain 2 echo-cancel coverage 32 echo-cancel suppressor timeouts interdigit 3!dial-peer voice 9 pots application session destination-pattern 0. no digit-strip direct-inward-dial port 0:D!dial-peer voice 99 voip destination-pattern [1-9]...T translate-outgoing calling 5 translate-outgoing called 1 voice-class codec 1 session protocol sipv2 session target dns:my.sipserver.com dtmf-relay h245-signal no call fallback no vad!gateway timer receive-rtcp 5!sip-ua retry invite 2 retry response 2 retry bye 2 retry cancel 2-------------- next part --------------A non-text attachment was scrubbed...Name: isndlog_sig.txtType: plain/textSize: 4876 bytesDesc: not availableUrl : https://puck.nether.net/pipermail/cisco-voip/attachments/20040512/48048430/… --------------------------------- Previous message: [cisco-voip] CCM 3.2 and call Parks.. Next message: [cisco-voip] Domain name Change Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] --------------------------------- More information about the cisco-voip mailing list __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers --------------------------------- Do you Yahoo!? Yahoo! Small Business - Try our new resources site! _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 5 months

[Serusers] Undefined symbol: fm_malloc when load pa module

by Cameron Beattie

I am attempting to load the pa module on SER 0.9.0 and am getting the message could not open module /usr/local/lib/ser/modules/pa.so undefined symbol: fm_malloc. I have seen others advise reinstalling when getting this message. I have done a make proper; make; make install but am still getting the error. Any suggestions? Regards Cameron

19 years, 5 months

[Serusers] call fwd all with avp_ops question

by Steve Blair

I'm trying to use avp_ops to implement call forward all but it is not working. I'm hoping this list can offer some suggestions. The idea is that a use turns call forward all on or off via the web page. This sets the flag callfwdall to y or n. If flag=y then the user must enter a phone number to which all calls will be forwarded. When a calls arrives for the users phone number callfwdall is checked, if = y then fwdaddr is written to r-uri and the call proceeds through the rest of the route blocks. This is so that any ACL checks can be done on the newly written r-uri so that user cannot forward to an otherwise inaccessible number. Well this isn't working. Here is a snippet from my config. All other calling is working fine. if (method=="INVITE") { if (lookup("location") | lookup("aliases") ) { xlog("L_INFO", "\n[SER]: AVP LOAD: Checking callfwdall flag \n\n"); if (avp_db_load("$ruri/username", "s:callfwdall")) { if (avp_check("s:callfwdall","eq/y/i")) { if (avp_db_load("$ruri/username", "s:fwdaaddr")) { avp_write("$ruri/username", "s:ocn"); avp_pushto("$ruri/username", "s:fwdaaddr"); rewritehostport("net.isc.upenn.edu:5060"); xlog("L_INFO", "\n[SER]: AVP LOAD new ruri: Time: [%Tf] From uri <%fu> To < %tu> Method: <%rm> R-uri: <%ru> Contact Header: <%ct> \n\n"); setflag(3); } else { xlog("L_INFO", "\n[SER]: AVP LOAD: no fwdaaddr \n\n"); }; Thanks,Steve

19 years, 5 months

[Serusers] sems 0.8.12 & ser 0.9.0 install-dir is broken

by Barry Murphy

Hi, During the make of sems I receive this error and I fully understand why, because there is no sems in the local folder. This is after a gmake , gmake install, everything else installs fine, but what use is the other stuff if the sems binary isnt compiled touch /usr/local/sbin/sems install -m 755 sems /usr/local/sbin/ install: sems: No such file or directory gmake: *** [install-bin] Error 71 root@max:/usr/home/ser/ser/answer_machine# uname -a FreeBSD max.unix.co.nz 4.10-RELEASE-p3 FreeBSD 4.10-RELEASE-p3 #1: Wed Jan 12 13:20:12 NZDT 2005 icepick@max.unix.co.nz:/usr/src/sys/compile/MAX i386 Latest sems: cvs co -r ser_rel_0_8_12 answer_machine Latest ser: version: ser 0.9.1 (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_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 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 19:38:23 Mar 16 2005 with gcc 2.95 Thanks Barry

19 years, 5 months

[Serusers] CALL-ID

by Kapil Dhawan

Hi All When CALL-ID is considered to be a globally unique identifier, then why do we describe one dialog with a combination of CALL-ID, TO tag and FROM tag. Regards _________________________________________________________________ News, views and gossip. http://www.msn.co.in/Cinema/ Get it all at MSN Cinema!

19 years, 5 months

RE: [Serusers] Mediaproxy still consuming a lot of RAM resources.

by Ricardo Martinez

Hello list. Regarding to this issue. These are the results for my "production" machine after upgrade the Python from 2.3 to 2.4.1. Since Friday the %MEM is 0.5, as the output of the "top" command indicate : PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 6131 root 15 0 5460 5460 2492 S 9.9 0.5 119:04 0 mediaproxy.py So, it works!. For the record i'm using : OS : Red Hat 9.0 Python : 2.4.1 SER: 0.8.14 mediaproxy : 1.2.1 Regards, Ricardo Martinez.- > -----Mensaje original----- > De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Enviado el: Viernes, 08 de Abril de 2005 18:18 > Para: 'Lucas Aimaretto'; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] Mediaproxy still consuming a lot of > RAM resources. > > > Regarding to this issue > Well, thanks to Lucas, just to test i changed the > pyhton version in > my "lab" server from python2.3 to pyhton2.4.1. I made the > same test with > the rtpgenerator and "voila". > > Check it out : > > Before 80 simultaneus calls : > > PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME COMMAND > 13915 root 15 0 4628 4628 2068 S 1.0 0.9 0:00 > mediaproxy.py > > During the test : > > PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME COMMAND > 13915 root 25 0 4864 4864 2068 R 99.9 0.9 0:31 > mediaproxy.py > > After the test : > > PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME COMMAND > 3356 root 15 0 1600 1600 1036 S 0.0 0.3 0:00 bash > 13915 root 15 0 4864 4864 2068 S 0.0 0.9 0:44 > mediaproxy.py > > 0.9 % !!!!! > > I made the test a couples of times later and the %MEM used is > still 0.9%. > I guess that it was maybe a problem between the python2.3 and the > mediaproxy? Don't know... > > I'm going to change the python version in my production > machine. i'm going > to keep you inform about the results... > > Thanks to Lucas...again. > > Best Regards, > > Ricardo Martinez.- > > > -----Mensaje original----- > > De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > > Enviado el: Viernes, 08 de Abril de 2005 17:33 > > Para: 'Lucas Aimaretto'; serusers(a)lists.iptel.org > > Asunto: RE: [Serusers] Mediaproxy still consuming a lot of > > RAM resources. > > > > > > For example now i have 14 simultaneus calls. > > > > PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME > > CPU COMMAND > > 11760 root 15 0 601M 518M 2396 R 7.2 > 51.4 > > 69:10 0 mediaproxy.py > > > > > > When the active calls are 0 again the CPU % reach 0.2%, but > > the %MEM is > > greater than 51.4%, and always increasing.. > > A couple of questions. > > How you launch mediaproxy?. And in your last mail what are > the pyhton > > proccess running.. i don't see the mediaproxy process running. > > Thanks again. > > > > Ricardo.- > > > > > -----Mensaje original----- > > > De: Lucas Aimaretto [mailto:lucas@cyneric.com] > > > Enviado el: Viernes, 08 de Abril de 2005 17:20 > > > Para: 'Ricardo Martinez'; serusers(a)lists.iptel.org > > > Asunto: RE: [Serusers] Mediaproxy still consuming a lot of > > > RAM resources. > > > > > > > > > > But when you check the use of RAM memory in your machine... > > > > (maybe with the command "top" or the command "free") don't > > > > you notice a increasing amount of RAM used without being > > > > released? As i mentioned i have this problem.. this is a > > > > snippet of the output of the "top" command in my machine: > > > > > > > > PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME > > > > CPU COMMAND > > > > 11760 root 15 0 590M 516M 2396 S 0.9 > > > 51.2 > > > > 68:15 0 mediaproxy.py > > > > > > > > 51.2% of memory used? What version of pyhton are you using?, > > > > what OS are you using?. Thanks! > > > > > > Hey!, how many calls where you proxying when you had that usage of > > > memmory ? > > > > > > Regards, > > > > > > Lucas > > > > > > -- > > > No virus found in this outgoing message. > > > Checked by AVG Anti-Virus. > > > Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: > > 07/04/2005 > > > > > > > > > > _______________________________________________ > > 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 >

19 years, 5 months

[Serusers] SER Calls not working with Alias

by Mr Greg Plater

Hi I am having trouble with calls. I have three clients created with alias's asterisk = 38212352, user1 = 38212351 , user2 =38212350 All users can register and I can see with serctl ul show the aliais and the user names. I am using MYSQL and SERWEB also. When I try to call the alias from each user 1 & 2 nothing happens. I think I have to correct something in the route section. I want users to be able to call by alias and calls for PSTN to route to Asterisk with 0 prefix. I start ser with /usr/local/sbin/ser -D -E Can someone please help put my brain to rest. Thank you for all your support. # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.1.4", "subscriber")) { www_challenge("192.168.1.4", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; if (method == "INVITE" && (uri=~"^sip:0")){ rewritehostport ("192.168.1.5:5060"); t_relay(); break(); }

19 years, 5 months

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sr-users April 2005