sr-users April 2005

sr-users@lists.kamailio.org

250 participants
697 discussions

by Christian de Balorre

Hi, Anyone knows how to handle 487 Request Terminated on callee side when it recieves a "CANCEL" ? It triggers the failure_route logic bloc and I would like to avoid this. ser.sfg route [3] { # Local call, voip-voip, if (uri =~"^sip:7[0-9][0-9]@.*") { rewritehost("neotelecoms.fr"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; t_on_failure("1"); t_relay(); break; failure_route[1] { if (method == CANCEL) { break; }; revert_uri(); rewritehostport("192.168.10.122:5060"); prefix("7"); append_branch(); t_relay(); break; Christian de BALORRE Ingénieur Systèmes & Réseaux Neotelecoms - MFN France 7-9, rue Petit 92582 Clichy Cedex Tel 01 49 97 07 35

19 years, 6 months

Re [Serusers] RTPProxy only fails for Public to Private communication

by Vivienne Curran

Hi Greger (Sorry forgot to copy the serusers mailing list the 1st time), I made that change and my on_reply_route now looks like this: onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { fix_nated_contact(); if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); }; } else if (nat_uac_test("1")) { fix_nated_contact(); }; } Unfortunately when I restarted rtpproxy, SER and reregistered my phones, I still had the same message dump. U 84.203.148.146:5060 -> 157.190.74.151:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd 17ddd1b59ead49..From: "2092" <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c ..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b218 70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0. 5.18..Contact: <sip:2093@172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY, REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: rep laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP 0..a=sendrecv.. a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes.. Any more thoughts? BR Viv "Greger V. Teigre" <greger(a)teigre.com> wrote: Look at this: U 84.203.148.146:5060 -> 157.190.74.151:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd 17ddd1b59ead49..From: "2092" <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c ..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b218 70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0. 5.18..Contact: <sip:2093@172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY, REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: rep laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP 0..a=sendrecv.. a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes.. I assume this is where you get an error message. You haven't called fix_nated_contact() for this message, and in fact I believe there may be an error in the ONsip.org example where a line has been lost. 192. onreply_route[1] { 193. 194. if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { 195. if (!search("^Content-Length:\ 0")) { 196. force_rtp_proxy(); 197. }; 198. } else if (nat_uac_test("1")) { 199. fix_nated_contact(); 200. }; 201. } fix_nated_contact() should go in between line 194 and 195. Could you please confirm that this works? I will look at the config file. g-( ---- Original Message ---- From: Vivienne Curran To: serusers(a)lists.iptel.org ; greger(a)teigre.com Sent: Wednesday, April 06, 2005 04:12 PM Subject: Re: RTPProxy fails only for Private to Public communication > Just as an extra : I have a sniff of the message for when a public > client (2092)rings a private client (2093)included at the bottom of > this email. I cant see anything wrong with them but maybe it will > shed more light on the matter. > > Vivienne Curran <vivcurran(a)yahoo.co.uk> wrote: > I changed the line modparam("nathelper", "rtpproxy_sock", > "/var/run/rtpproxy.sock") to modparam("nathelper", "rtpproxy_sock", > "udp:localhost:22222") and started the rtpproxy as ./rtpproxy -s udp > from the relevant directory and this resulted in a series of > "rtpp_command: no response from rtpproxy" and rtpproxy temporarily > disabled" errors. If I return to the original modparam and start it > as ./rtpproxy then it works but like I said when the private client > rings the public client, I get "ERROR: send_rtpp_command: cant read > reply from a RTP Proxy". > > Any further ideas? Has anyone on the mailing list experienced this? I > am using the script given in the onsip getting started doc for 0.9.0. > but am using ser 0.8.14. > > BR, > Vivienne Send instant messages to your online friends http://uk.messenger.yahoo.com

19 years, 6 months

[Serusers] permissions module allow_routing

by Glenn Dalgliesh

Sorry to post again but I am really kind stuck on this one. I am having trouble with this module. I am trying to deny calls but the only command that seems to effect calls in the permissions.deny file is ALL:ALL. I am not sure if this is a problem with my syntax or other. Below is the relevant info. Thanks modparam("permissions", "default_allow_file", "/usr/local/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/usr/local/etc/ser/permissions.deny") if (lookup("location")) { log(1, "LOG: Match location\n"); append_hf("P-hint: usrloc applied\r\n"); if (allow_routing()) { log(1, "LOG: Allow_routing\n"); append_branch(); rewritehostport ("xxx.yyy.83.5:5060"); route(9); #t_relay(); break; } else { #append_branch(); log(1, "LOG: Post allow route failed voicemail\n"); rewritehostport ("xxx.yyy.83.5:5060"); prefix("2"); route(9); break; }; }; /usr/local/etc/ser/permissions.allow - is empty /usr/local/etc/ser/permissions.deny - various attempts "^sip:4103539999@xxx\.yyy\.88\.16$" : "^sip:5505@xxx\.yyy\.83\.4$" ALL : "^sip:5505$" ALL : "^sip:5505@xxx\.yyy\.83\.4$" "^sip:4103539999$" : ALL Ngrep of INIVTE to SER INVITE sip:5505@xxx.yyy.83.4 SIP/2.0..Via: SIP/2.0/UDP xxx.yyy.88.16:5060;branch=z9hG4bK7325aa3b..From: "Cell Phone MD" <sip:41 bd19a3b299ac6605833f7@xxx.yyy.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Thu, 07 Apr 2005 18:22:03 GMT..Allow: INV 159.88.16..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.1 (i386/linux))..Content-Length: 0..Warning: 392 xxx.yyy.83.4:5060 INVITE sip:5505@xxx.yyy.83.5:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:xxx.yyy.83.4;ftag=as424541a9;lr=on>..Via: SIP/2.0 57cdeb7c798bd19a3b299ac6605833f7@xxx.yyy.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Thu, 07 Apr 2005 18:22:03 GMT. .Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 238..P-hint: usrloc applied... INVITE sip:5505@69.250.205.31:5068 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:xxx.yyy.83.4;ftag=as424541a9;lr=on>..Via: SIP/2 : 57cdeb7c798bd19a3b299ac6605833f7@xxx.yyy.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Thu, 07 Apr 2005 18:22:03 GM T..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 238..P-hint: usrloc applied. -ID: 57cdeb7c798bd19a3b299ac6605833f7@xxx.yyy.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPT 798bd19a3b299ac6605833f7@xxx.yyy.88.16..CSeq: 102 INVITE..User-Agent: Grandstream BT100 1.0.5.16..Content-Length: 0.... _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 6 months

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

by Ricardo Martinez

Hello List. Any comments here? Thanks! Ricardo Martinez.- > -----Mensaje original----- > De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Enviado el: Jueves, 07 de Abril de 2005 13:07 > Para: 'serusers(a)lists.iptel.org' > Asunto: [Serusers] Mediaproxy still consuming a lot of RAM resources. > > > Hello Again. > A few weeks ago i posted about my mediaproxy consuming > RAM resources > without releasing it. I upgraded to mediaproxy version 1.2.1 > and even i > modified my mediaproxy configuration (in ser.cfg) with some > recommendation > from Paul (Java Rockx). All these efforts were useless, my > mediaproxy is > still consuming RAM and not releasing it. I made a simple > test.. Without > SER running i used the rtpgenerator (in the utils directory > of mediaproxy) > to load the mediaproxy, and this is what i found. > At the beggining of my test the "free" command show me this :: > > Every 1s: free > Thu Apr 7 13:50:05 2005 > > total used free shared > buffers cached > Mem: 514196 429208 84988 0 > 73672 217444 > -/+ buffers/cache: 138092 376104 > Swap: 1044216 0 1044216 > > 84 M free. > > I run the rtp proxy for about 80 simmultaneus calls. > > When i finish the rtp sessions i see the output of free command: > Every 1s: free > Thu Apr 7 13:53:43 2005 > > total used free shared > buffers cached > Mem: 514196 437868 76328 0 > 74952 216288 > -/+ buffers/cache: 146628 367568 > Swap: 1044216 0 1044216 > > 76 M free. > > As you can see it seems that the mediaproxy is not releasing > the RAM memory. > I see the command "top" and this is what i get : > > PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME COMMAND > 26972 root 15 0 12320 12M 2004 S 0.0 2.3 1:02 > mediaproxy.py > > The 2.3% is always increasing. > > Can someone help me here? > Thanks! > > Ricardo Martinez.- > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

19 years, 6 months

[Serusers] symmtric NAT setup on LINUX

by anand kishore p

Hi I would like to know can we configure the symmetric NAT on a linux machine , is there any scripts available for those. If any one have anyidea about the different NAT configurations on LINUX machine please let me know. thanks, kishore

19 years, 6 months

[Serusers] SER latest version

by Llanos Serna García-Conde

19 years, 6 months

[Serusers] Starting SER : PID file exists (/var/run/ser.pid) already running?

by Aisling O'Driscoll

Hi, I am wondering if anyone has ever come across this error before?: Starting SER : PID file exists! (/var/run/ser.pid) already running? I am using ser 0.9.0 and it worked like a charm the first few times that I stopped and started SER. There are no errors in /var/log/messages. I also started SER as /opt/ser/sbin/ser -c and it showed that the config was ok and exited. However when I check pstree, I dont see SER listening. Could it perhaps be something got to do with the database?. Example: Should the version table in the SER database contain "version" or "table_version"? I have tried dropping and restarting the database i.e. /opt/ser/sbin/ser_mysql.sh drop, then ./opt/ser/sbin/ser_mysql.sh create. Any ideas would be appreciated. Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

19 years, 6 months

[Serusers] Hi All

by Kapil Dhawan

Hi All I am a regular visitor of your mailing list. I am a 'C' developer with no background knowledge of SIP related protocols, but always wanted to work in it. Could you please guide me, how should i start it. Which RFC's should i read first and etc. Regards _________________________________________________________________ Expressions unlimited! http://server1.msn.co.in/sp04/messenger/ The all new MSN Messenger!

19 years, 6 months

[Serusers] [Sems] SER 0.9.0 + SEMS

by Nicolas Ruiz

HI ! I have a problem, My PSTN GATEWAY is a Cisco AS5300 with codec g729r8. When I Receive an incomming call, I have this error when I forward to sems : Sems[21873]: Error: (AmSession.cpp)(run)(193): 606 could not find compatible payload But If I set: codec gsmfr, it's good, but after, I can't receive calls, because my phones are Codec g729 !!! How can I use voicemail and ser with 2 codecs??? thanks a lot Best Regards Nicolas RUIZ -----Message d'origine----- De : Raphael Coeffic [mailto:coeffic@fokus.fraunhofer.de] Envoyé : mardi 5 avril 2005 11:51 À : Nicolas Ruiz; sems(a)lists.iptel.org Objet : Re: [Sems] SER 0.9.0 + SEMS > Does SEMS works correctly with SER 0.9.0 ? Yes. > Because I use the TM module and I have this error : > > Apr 4 16:32:25 sip2 Sems[8474]: Error: 404 voicemail: no email address > for > user <0170708694> > Apr 4 16:32:25 sip2 ser[11211]: ERROR: fifo_server: command vm_reply is > not > available It looks like you are trying to use SEMS 0.8.12 (ser_rel_0_8_12) together with SER 0.9.0. You should use the head version of SEMS (no tag). This version will become the new stable version. -Raphael. > Best Regards > > Nicolas RUIZ > FRANCE, Paris

19 years, 6 months

[Serusers] Re [cisco-voip] No early media for ISDN->SIP->ISDN

by karl

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

19 years, 6 months

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sr-users April 2005