sr-users October 2008

sr-users@lists.kamailio.org

93 participants
169 discussions

Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process

by Daniel Cardoso Alves

Hi Henning Westerholt, Thanks for you reply... But I would like to know if there is a number that improve the performance... For example: If come 100 INVITE at the same moment, is there a NUMBER OF CHILDREN that has the best perfomance? The number of children depends on something? Thanks --- Em qua, 29/10/08, Henning Westerholt <henning.westerholt(a)1und1.de> escreveu: De: Henning Westerholt <henning.westerholt(a)1und1.de> Assunto: Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process Para: users(a)lists.kamailio.org Cc: dcardosoa(a)yahoo.com.br Data: Quarta-feira, 29 de Outubro de 2008, 11:25 On Wednesday 29 October 2008, Daniel-Constantin Mierla wrote: > > Is there a number of children that the Kamailio/Openser for improve > > the performance? > > > > Each process takes care of a request? > > yes, each process takes care of a SIP message (request/reply). Hi Daniel Cardoso, take a look to the core cookbook in our wiki [1] for more informations how you can control the number of childrens per protocol (UDP, TCP..). Cheers, Henning [1] http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#children Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

16 years

[OpenSER-Users] Openser1.3.2 + snmp

by Ricardo Alves Ferreira Silva

Hi all, Trying to install Openser1.3.2.... everything is OK and fine... but when I try to add support for snmp there are errors.. Dig dep into google, recheck all dependencies and didn't succeed... My env is FreeBSD 7.0-STABLE i386 The full net-snmp package is installed: atila# pkg_info | grep net-snmp net-snmp-5.3.2_3 An extendable SNMP implementation These are the error messages.... make modules/libsnmpstats modules.... Compiling alarm_checks.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -DMOD_NAME='"snmpstats"' -DNAME='"openser"' -DVERSION='"1.3.2-notls"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 4.2.1"' -D__CPU_i386 -D__OS_freebsd -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_NETINET_IN_SYSTM -DHAVE_KQUEUE -DHAVE_SELECT -c alarm_checks.c -o alarm_checks.o In file included from alarm_checks.c:38: openserObjects.h:41:38: error: net-snmp/net-snmp-config.h: No such file or directory openserObjects.h:42:40: error: net-snmp/net-snmp-includes.h: No such file or directory openserObjects.h:43:52: error: net-snmp/agent/net-snmp-agent-includes.h: No such file or directory If I edit each file and hardcode the path ... /usr/local/include/net-snmp/... there are no errors... What am I missing in making sure the path to the include files are OK... Tried a couple options but didn't succeed... tehre are tons of include files into the src.... Regards Ricardo Ferreira Yes, we do run Openser www.vipway.com.br Enterprise VoIP Services

16 years

[Serusers] Error in jabber module - XJAB:extract_aor: Error while parsing URI

by Tomasz Zieleniewski

Hi, We are using ser 2.0 We encountered the following error: 7(407) route[PRESENCE_SUBSCRIBE](SUBSCRIBE cseq:933): (handle_subscription(jabber)) 7(407) XJAB:extract_aor: Error while parsing URI But still SER sends an 200 OK response and after then a NOTIFY. Here is the message received by ser: SUBSCRIBE sip:lkl%touk.pl@jabber.rd.touk.pl SIP/2.0 Via: SIP/2.0/UDP 192.168.0.165;rport;branch=z9hG4bKxakietyi Max-Forwards: 70 Proxy-Authorization: Digest username="tzl",realm="touk.pl",nonce="490987cb2702bb9db8d631e016825ad1c9cd12ac",uri="sip:lkl%touk.pl@jabber.rd.touk.pl",response="bd4474cf8af3d1ec8a7246872c0ecb46",algorithm=MD5,cnonce="mk57n39293",qop=auth,nc=00000001 To: <sip:lkl%touk.pl@jabber.rd.touk.pl> From: "Tomasz Zieleniewski" <sip:tzl@sip.rd.touk.pl>;tag=cejrf Call-ID: vxiulpsynbnidjo(a)sen.rd.touk.pl CSeq: 933 SUBSCRIBE Contact: <sip:tzl@192.168.0.165> Accept: application/pidf+xml Event: presence Expires: 3600 User-Agent: Twinkle/1.2 Content-Length: 0 Jabber config: # -- jabber params --- modparam("jabber", "jaddress", "touk.pl") modparam("jabber", "jport", 5222) modparam("jabber", "registrar", "serimgw(a)sip.rd.touk.pl") modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.rd.touk.pl=%") Please give us a hint what could be wrong. Kind regards Tomasz

16 years

[Serusers] Configuring SER as an outbound proxy for multiple Ekigas on a LAN

by Enrico Maria Crisostomo

Hi all. I'm a first-time would-be user of SER and I have the following problem: I was planning to use Ekiga on a LAN with several workstations and I was looking for documentation about the options I have and the best setup I can achieve. How to solve the problem of NAT traversal for multiple Ekigas was answered by Ekiga documentation: use a SIP proxy. The workstations are all equipped with Solaris 10 and I have a small Solaris 10 server too, so I thought it was natural for me to use SER as a SIP proxy. Unfortunately I'm finding no documentation about how to configure it as a simple outbound proxy and to say the truth I found very little documentation about configuring a SIP proxy in general. My idea would be just having SER behave as an outbound SIP proxy and use ekiga.net accounts for everybody (no need for a PBX as Asterisk). It seems a kind of basic configuration but I have no idea where to start from. Do you have suggestions and pointers to documentation to help me achieve this goal? Thank you in advance, Enrico. -- Ελευθερία ή θάνατος GPG key: 1024D/FD2229AF fpr: 9E07 D40E 33A5 5993 6FC5 09A8 5BCF B1F2 FD22 29AF

16 years

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by Ricky Gutierrez

Hi luzano thank you for your help and time, this it is my full ngrep. I have asterisk, mediaproxy and openser together in the same pc I have the doubt that when a incoming call from the PSTN by asterisk, sends it to an extension of openser, to which I do not request authentication to him within invites, I believe that there it is where I have problems with mediaproxy that it leaves you want to see of the openser.cfg, or everything? regards .. interface: any filter: (ip) and ( port 5060 ) # U +0.063740 192.168.10.1:5070 -> 192.168.10.1:5060 INVITE sip:113@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 14750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062 INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Content-Length: 0 # U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.000014 192.168.1.64:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.956449 192.168.10.19:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000027 192.168.1.64:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.156145 190.184.35.4:5060 -> 192.168.1.64:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000557 192.168.10.1:5070 -> 192.168.10.1:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 # U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 P-hint: LR|fixcontact,setflag6 # U +0.886545 192.168.10.28:5060 -> 192.168.10.1:5060 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> To: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:12:29 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango.

16 years

[Kamailio-Users] kamailio and rtpproxy-no audio

by Fabian Borot

hello I am trying to use rtpproxy and kamailio. Below please find my kamailio.cfg file. But this is what is happening: 1- ext 101[IP 76.109.183.2] is able to send audio to kamailio's IP [65.111.185.187]but it does not reach ext 100 IP [76.109.15.75]. 2- ext 100 does not send RTP at all. 3- I still see the private IPs in the SDPs of both exts. I am aattaching a wireshark capture Thank you in advance Fabian # $Id: kamailio.cfg 4576 2008-08-05 16:22:38Z klaus_darilion $ # # Kamailion basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = '/usr/local/etc/kamailio/tls/user/user-cert.pem' #tls_private_key = '/usr/local/etc/kamailio/tls/user/user-privkey.pem' #tls_ca_list = '/usr/local/etc/kamailio/tls/user/user-calist.pem' port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:65.111.185.187:5060 ####### Modules Section ######## #set module path mpath='/usr/local/lib/kamailio/modules/' /* uncomment next line for MySQL DB support */ loadmodule 'db_mysql.so' loadmodule 'sl.so' loadmodule 'tm.so' loadmodule 'rr.so' loadmodule 'maxfwd.so' loadmodule 'usrloc.so' loadmodule 'registrar.so' loadmodule 'textops.so' loadmodule 'mi_fifo.so' loadmodule 'uri_db.so' loadmodule 'uri.so' loadmodule 'xlog.so' loadmodule 'acc.so' /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule 'auth.so' loadmodule 'auth_db.so' /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'alias_db.so' /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see 'multi-module params' section ) */ #loadmodule 'domain.so' /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'presence.so' #loadmodule 'presence_xml.so' loadmodule 'nathelper.so' # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam('mi_fifo', 'fifo_name', '/tmp/kamailio_fifo') # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam('rr', 'enable_full_lr', 1) # do not append from tag to the RR (no need for this script) modparam('rr', 'append_fromtag', 0) # ----- rr params ----- modparam('registrar', 'method_filtering', 1) /* uncomment the next line to disable parallel forking via location */ # modparam('registrar', 'append_branches', 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam('registrar', 'max_contacts', 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam('uri_db', 'use_uri_table', 0) modparam('uri_db', 'db_url', '') # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam('acc', 'early_media', 1) modparam('acc', 'report_ack', 1) modparam('acc', 'report_cancels', 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable 'append_fromtag' in 'rr' module */ modparam('acc', 'detect_direction', 0) /* account triggers (flags) */ modparam('acc', 'failed_transaction_flag', 3) modparam('acc', 'log_flag', 1) modparam('acc', 'log_missed_flag', 2) /* uncomment the following lines to enable DB accounting also */ modparam('acc', 'db_flag', 1) modparam('acc', 'db_missed_flag', 2) # ----- usrloc params ----- modparam('usrloc', 'db_mode', 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam('usrloc', 'db_mode', 2) #modparam('usrloc', 'db_url', # 'mysql://openser:openserrw@localhost/openser') modparam('usrloc', 'nat_bflag', 3) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam('auth_db', 'calculate_ha1', yes) modparam('auth_db', 'password_column', 'password') modparam('auth_db', 'db_url', 'mysql://openser:openserrw@localhost/openser') modparam('auth_db', 'load_credentials', '') # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam('alias_db', 'db_url', # 'mysql://openser:openserrw@localhost/openser') # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam('domain', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('domain', 'db_mode', 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam('alias_db|auth_db|usrloc|uri_db', 'use_domain', 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam('presence|presence_xml', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('presence_xml', 'force_active', 1) #modparam('presence', 'server_address', 'sip:192.168.1.2:5060') # ----- nathelper params ----- modparam('nathelper', 'natping_interval', 10) modparam('nathelper', 'ping_nated_only', 1) modparam('nathelper', 'rtpproxy_sock', 'udp:localhost:8899') modparam('nathelper', 'rtpproxy_disable_tout', 20) modparam('nathelper', 'rtpproxy_tout', 2) modparam('nathelper', 'rtpproxy_retr', 2) modparam('nathelper', 'sipping_from', 'sip:pinger@mykamailio.com') modparam('nathelper', 'sipping_bflag', 7) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header('10')) { sl_send_reply('483','Too Many Hops'); exit; } if (is_method('REGISTER')) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize('', 'subscriber')) { www_challenge('', '0'); exit; } if (!save('location')) sl_reply_error(); exit; } if (!lookup('location')) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply('404', 'Not Found'); exit; case -2: sl_send_reply('405', 'Method Not Allowed'); exit; } } # when routing via usrloc, log the missed calls also setflag(2); # record routing if (!is_method('REGISTER|MESSAGE')) { record_route(); } # account only INVITEs if (is_method('INVITE')) { setflag(1); # do accounting } # for INVITEs enable some additional helper routes if (is_method('INVITE')) { force_rtp_proxy(); } if (!t_relay()) { sl_reply_error(); }; exit; # CANCEL processing if (is_method('CANCEL|BYE')) { unforce_rtp_proxy(); if (t_check_trans()) t_relay(); exit; } } _________________________________________________________________ Stay organized with simple drag and drop from Windows Live Hotmail. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_102008

16 years

[Kamailio-Users] version cdrtool mediaproxy 1.9.1

by Ricky Gutierrez

Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

16 years

Re: [Kamailio-Users] REGISTER port with Xlite 2.0

by ingdavidcespedes＠cable.net.co

Hi I download Twinkle, and install all the prerequisites, except for Qt. I have some problems with it. First I download Qt for Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't know if it is the one I have to download, because In the Twinkle page, it only says you have to install "Qt for open source (3.3.0 or later, but not Qt4)". Then I do all the steps cd /tmp gunzip qt-x11-opensource-src-4.4.3.tar.gz # uncompress the archive tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it cd /tmp/qt-x11-opensource-src-4.4.3 ./configure #It took more than 6 hours make su -c "make install" Then I retry to install Twinkle entering in Twinkle directory an doing ./configure but I receive the following error: checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. So I read the FAQ of Twinke (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that I can skip this error by doing ./configure --disable-qt-check, but now I receive the following error: checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. So I realize that I didn't have libqt-mt so I try to recompile QT with "./configure -thread" but it didn't let me, so I try with "./configure -DQT_THREAD_SUPPORT" and the compilation took also mother tan 6 hours, and I retry to install Twinkle, but without success, I receive the same message. checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. I don't Know what I am doing wrong. Any Help!!!!! please. David Cespedes. ----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Martes, Octubre 28, 2008 8:15 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > is twinkle able now to display rich presence status? Last time I > tried> was showing only online/offline status. > > For now just online/offline status. > > But it has other great features as: > - Indication of Message Composition for Instant Messaging - RFC 3994 > - PingPong TCP keepalive. > - TCP/UDP > - PRACK > - q-value in registration. > > For me it's the most reliable softphone. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

16 years

Re: [Kamailio-Users] REGISTER port with Xlite 2.0

by ingdavidcespedes＠cable.net.co

Excuse me list, it was an error copy to this list, but maybe some one can help me. ----- Mensaje original ----- De: Alex Balashov <abalashov(a)evaristesys.com> Fecha: Miércoles, Octubre 29, 2008 11:31 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > This is not the Twinkle list. > > ingdavidcespedes(a)cable.net.co wrote: > > > Hi > > > > I download Twinkle, and install all the prerequisites, except for > Qt. I have some problems with it. First I download Qt for > Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't > know if it is the one I have to download, because In the Twinkle > page, it only says you have to install "Qt for open source (3.3.0 > or later, but not Qt4)". Then I do all the steps > > > > cd /tmp > > gunzip qt-x11-opensource-src-4.4.3.tar.gz # > uncompress the archive > > tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it > > cd /tmp/qt-x11-opensource-src-4.4.3 > > ./configure #It took more than 6 > hours> make > > su -c "make install" > > Then I retry to install Twinkle entering in Twinkle directory an > doing ./configure but I receive the following error: > > > > checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not > found in the pkg-config search path. Perhaps you should add the > directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment > variable No package 'qt-mt' found Package qt-mt was not found in > the pkg-config search path. Perhaps you should add the directory > containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable > No package 'qt-mt' found > > configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < > 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment > variable if your libraries are in a nonstandard prefix so pkg- > config can find them. > > > > So I read the FAQ of Twinke > (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that > I can skip this error by doing ./configure --disable-qt-check, but > now I receive the following error: > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > So I realize that I didn't have libqt-mt so I try to recompile QT > with "./configure -thread" but it didn't let me, so I try with > "./configure -DQT_THREAD_SUPPORT" and the compilation took also > mother tan 6 hours, and I retry to install Twinkle, but without > success, I receive the same message. > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > I don't Know what I am doing wrong. Any Help!!!!! please. > > > > David Cespedes. > > > > ----- Mensaje original ----- > > De: Iñaki Baz Castillo <ibc(a)aliax.net> > > Fecha: Martes, Octubre 28, 2008 8:15 am > > Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > > > >> 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >>> is twinkle able now to display rich presence status? Last time > I > >> tried> was showing only online/offline status. > >> > >> For now just online/offline status. > >> > >> But it has other great features as: > >> - Indication of Message Composition for Instant Messaging - RFC > 3994>> - PingPong TCP keepalive. > >> - TCP/UDP > >> - PRACK > >> - q-value in registration. > >> > >> For me it's the most reliable softphone. > >> > >> > >> -- > >> Iñaki Baz Castillo > >> <ibc(a)aliax.net> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

16 years

[Kamailio-Users] Fwd: How to enable local_route feature ?

by Pascal Maugeri

Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal ---------- Forwarded message ---------- From: Pascal Maugeri <pascal.maugeri(a)gmail.com> Date: Wed, Oct 29, 2008 at 11:17 AM Subject: How to enable local_route feature ? To: Henning Westerholt <henning.westerholt(a)1und1.de> Cc: devel(a)lists.openser.org, openser users <users(a)lists.openser.org> Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal On Thu, Jun 26, 2008 at 1:11 PM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > On Friday 06 June 2008, Bogdan-Andrei Iancu wrote: > > OpenSER has a new type of route: local route. This route is to be > > executed when a brand new request is internally generated (openser being > > the UAC) ; such requests are result of : > > - t_uac_dlg MI command > > - msilo, sms, presence, cpl-c, dialog (and others) modules > > [..] > > Hello all, > > the addition of this feature short before the code freeze caused some > conflicts, which could not resolved sufficently in time for this release. > Thus the OpenSER board was asked to decide in this matter. The majority of > the board voted for decativating this feature in the 1.4 release series. > > For more details please take a look at the developer mailing list at [1], > [2] > and [3]. Details about the "dectivate" commit can be found at [4] > > The local route functionality is still available, but its not activated in > the > default installation. In order to use this feature its necessary to set > the "USE_LOCAL_ROUTE" define in the "Makefile.defs" file and compile your > own > binary from the source. Then it will possible to use this feature. > > In the 1.5.0 release this functionality (or something resemble this) will > be > present, fully usable and supported. > > With best regards, > > Henning Westerholt > > > [1] http://lists.openser.org/pipermail/devel/2008-June/014028.html > [2] http://lists.openser.org/pipermail/devel/2008-June/014122.html > [3] http://lists.openser.org/pipermail/devel/2008-June/014204.html > [4] http://lists.openser.org/pipermail/devel/2008-June/014444.html > > > -- > Henning Westerholt - Development Consumer Products / DSL Core > 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

16 years

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sr-users October 2008