sr-users October 2008

sr-users@lists.kamailio.org

93 participants
170 discussions

[Kamailio-Users] Dimensioning a telephony system based on openser!
by Nuno Marques 30 Oct '08

30 Oct '08

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites: - 10000 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity; Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance, Nuno

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Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 30 Oct '08

30 Oct '08

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

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[OpenSER-Users] Openser1.3.2 + snmp
by Ricardo Alves Ferreira Silva 30 Oct '08

30 Oct '08

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

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[Serusers] Error in jabber module - XJAB:extract_aor: Error while parsing URI
by Tomasz Zieleniewski 30 Oct '08

30 Oct '08

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

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[Serusers] Configuring SER as an outbound proxy for multiple Ekigas on a LAN
by Enrico Maria Crisostomo 30 Oct '08

30 Oct '08

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

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 30 Oct '08

30 Oct '08

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.

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[Kamailio-Users] kamailio and rtpproxy-no audio
by Fabian Borot 30 Oct '08

30 Oct '08

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

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[Kamailio-Users] version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 29 Oct '08

29 Oct '08

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

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

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 >

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

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 >

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sr-users October 2008