- sr-users - kamailio.org

[Serusers] serweb php session
by Iqbal 15 Sep '05

15 Sep '05

Hi On one of my servers I'm using serweb with php session ID's is anyone else having a problem where sometimes the session is not created, hence login, or new user registrations cannot happen, cause it gives a 302 error Iqbal

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[Serusers] No VM module and IVR plugin
by Nicky 15 Sep '05

15 Sep '05

Dear all, Please help. I am trying to download Sip_router_rel_0_8_12 and Sems version ser_rel_0_8_12 from the CVS but in the sems there is missing module for IVR. Then I try to download 0.9.0 version for both but there is missing VM module in Ser which is required to link to Sems. Please advice if I would like to use 0.9.0 version, without VM module how can link to SEMS and why in the sems version ser_rel_0_8_12 the IVR plugin is missing and where can I find it. Thanks in advance.... Regards, Nicky

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[Serusers] Load balancer
by Kapil Dhawan 15 Sep '05

15 Sep '05

Dear List How can we implement a load balancer in SER. Regards _________________________________________________________________ Amitabh fan? Heres a chance to chat with the Big B himself. http://server1.msn.co.in/sp05/DJbk/index.asp Click here now!

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RE : [Serusers] Handoff to PSTN
by Olivier Taylor 15 Sep '05

15 Sep '05

what's on the line 143??? -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:46 À : 'Olivier Taylor' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] Handoff to PSTN The line 143 points to the script below. What modules will it be looking for? _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 2:33 PM To: 'Colin Jordan' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN please reply to all u have a module not loaded, the one involved on line 143 Olivier -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:28 À : 'Olivier Taylor' Objet : RE: [Serusers] Handoff to PSTN It gives me the following message: _______________________________________________________ Parse error (143,40-41): unknown command missing loadmodule? _______________________________________________________ _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 1:21 PM To: 'Colin Jordan'; serusers-request(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN just have a look at /var/log/messages u will find the line number where the error occurs Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Colin Jordan Envoyé : jeudi 15 septembre 2005 12:33 À : serusers-request(a)lists.iptel.org Cc : serusers(a)lists.iptel.org Objet : [Serusers] Handoff to PSTN I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~^sip:9[0-9]*@mydomain.com) { log(Forwarding to PSTN\n); t_relay_to( 192.168.0.2, 5060); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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RE : [Serusers] Handoff to PSTN
by Olivier Taylor 15 Sep '05

15 Sep '05

please reply to all u have a module not loaded, the one involved on line 143 Olivier -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:28 À : 'Olivier Taylor' Objet : RE: [Serusers] Handoff to PSTN It gives me the following message: _______________________________________________________ Parse error (143,40-41): unknown command missing loadmodule? _______________________________________________________ _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 1:21 PM To: 'Colin Jordan'; serusers-request(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN just have a look at /var/log/messages u will find the line number where the error occurs Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Colin Jordan Envoyé : jeudi 15 septembre 2005 12:33 À : serusers-request(a)lists.iptel.org Cc : serusers(a)lists.iptel.org Objet : [Serusers] Handoff to PSTN I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~^sip:9[0-9]*@mydomain.com) { log(Forwarding to PSTN\n); t_relay_to( 192.168.0.2, 5060); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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[Serusers] Handoff to PSTN
by Colin Jordan 15 Sep '05

15 Sep '05

I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~"^sip:9[0-9]*@mydomain.com") { log("Forwarding to PSTN\n"); t_relay_to( "192.168.0.2", "5060"); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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[Serusers] Handoff to PSTN
by Colin Jordan 15 Sep '05

15 Sep '05

I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~"^sip:9[0-9]*@mydomain.com") { log("Forwarding to PSTN\n"); t_relay_to( "192.168.0.2", "5060"); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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[Users] openser + osip
by Loide Mara 15 Sep '05

15 Sep '05

Hi, Did somebody implement a module to Openser that use libosip? I'm trying to implement it... Does somebody have some idea? Best regards -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

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[Serusers] SER with SEMS with Asterisk
by tdeillon＠cri74.org 15 Sep '05

15 Sep '05

Hello, I send this mail just to know if a configuration's documentation exist to configure them. Thanks, Thomas

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[Serusers] codec restrict mediaproxy.
by Vikash Khatuwala 15 Sep '05

15 Sep '05

Hi, Does anyone know if its possible to restrict the codecs in mediaproxy. ie. to force only lower bandwidth codecs to reduce the load. I couldnt find any info on this on the search engines. Thanks. Vikash.

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[Serusers] [about SER] mydomain is unresolveable
by 송 미진 15 Sep '05

15 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear Hi, serhelper. It’s my great pleasure to e-mail you. My name is Mi-Jin Song. I am a senior majoring in CS(Computer Science) at the Ewha Univ. in Korea. I just have a few questions about sip. First of all, I would like to explain about my situation. I am planning to make ‘testbed’ using 2 UA (User Agent)s and 3 sip servers, and I use linux fedora (version 3) as an operating system. In doing so, I have been using SER(Sip Express Router), sip open source provided by iptel.org, in order to use sip (session initiation protocol). When I was using sipsak, SER test tool made by your company, to check the network condition, but my computer did not seem to understand uri (sip:mi@sam.cse). For example, when I put root]#sipsak -T -s sip:mi@sam.cse I got an error message which is ‘sam.cse’ is unresolvable’ I made sure I put root]#export SIP_DOMAIN=”sam.cse” root]#serctl add mi 111 mi@sam. cse’ before. What I am guessing is that only using the order, export SIP _DOMAIN=“same. cse” does not do the uri setting. It’s been very hard since I could not find enough data and information regarding SER. I would like to know if you would be able to tell me what I should do to make the server recognize uri. If I need to fix something or use other methods additionally, could you please let me know? I will appreciate it very much if you could give an answer as soon as possible since the deadline for my project is coming very soon. Thank you so much. Sincerely, Mi-Jin Song. _________________________________________________________________ 보다 빠르고 보기 편한 뉴스. 오늘의 화제는 MSN 뉴스에서 확인하세요. http://www.msn.co.kr/news/

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[Serusers] Cisco 2600
by Ryan Pagquil 15 Sep '05

15 Sep '05

Hi Guys, Have anyone on the list tried setting up Cisco 2600 as SIP gateway together with SER? Could you explain what will be the function of the SIP gateway and it's relation with SER. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] Need sample incoming call acc data
by Eric Haskins 15 Sep '05

15 Sep '05

Hello, We are working on a billing solution for use with SER. We have completed a great PrePaid Billing solution for a client V1010.com but they currently only do outbound and arent planning to do inbound for a time. We are looking for some sample inbound call data from PSTN if anyone has some we would appreciate it! We can handle all Country codes and Area Code rates as well as custom rates as of now. So hope to have monthly billing completed soon. Thanks for any help Eric Haskins Lead Software Developer eric(a)rackspeed.net

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[Serusers] Re: Iqbal maybe you can help
by Ali Zaidi 14 Sep '05

14 Sep '05

Hi, Thanks for reply, problem is fixed i reinstalled rtp from source, it seems that every element in place, only thing left is to put ser on public NOT in DMZ. I tried Mediaproxy before doing rtpproxy and i gave up. i will come back to mediaproxy some time in future. Thanks, Ali On 9/14/05, Iqbal <iqbal(a)gigo.co.uk> wrote: > > Hi > > Unfortunately I dont usertpproxy, I use mediaproxy, which I find to be > better. > > Iqbal > > Ali Zaidi wrote: > > > Hi, > > > > I have trouble with RTPPRoxy. > > I downloaded a everything.GettingStarted.tar.gz package from Onsip, > > hello-world.3.04.cfg and auth-mysql.3.05.cfg work but when i use > > nat-rtpproxy.3.04.cfg i get following errors. > > > > [root@localhost root]# /etc/init.d/rtpproxy start > > [root@localhost root]# /etc/init.d/ser restart > > Stopping ser: [FAILED] > > Starting ser: [ OK ] > > [root@localhost root]# serctl ul show > > Dumping all contacts may take long: are you sure you want to proceed? > > [Y|N] y > > Error opening ser's FIFO /tmp/ser_fifo > > Make sure you have line fifo=/tmp/ser_fifo in your config > > [root@localhost root]# > > > > > > By the way i don't see rtpproxy.sock under \var\run\ > > > > Questions: > > > > If i use rtpptoxy, how we start it? i do "/etc/init.d/rtpproxy start" > > then it doesn't show if that is started or not unless i run "ps -A" > > Can we put SER into DMZ and do port forwarding OR SER must have Puplic > IP? > > If we can put SER on DMZ then what other ports need to be open beside > > 5060? > > Do i need new version of RTPProxy? > > > > > > > > # =============================================== > > > > debug=3 > > fork=yes > > log_stderror=no > > > > listen= x.x.x.x # put your server IP address here That is MY WAN IP. > > port=5060 > > children=4 > > > > dns=no > > rev_dns=no > > > > fifo="/tmp/ser_fifo" > > fifo_db_url=" mysql://ser:heslo@localhost/ser" > > > > loadmodule "/usr/local/lib/ser/modules/mysql.so" > > loadmodule "/usr/local/lib/ser/modules/sl.so" > > loadmodule "/usr/local/lib/ser/modules/tm.so" > > loadmodule "/usr/local/lib/ser/modules/rr.so" > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > > loadmodule "/usr/local/lib/ser/modules /usrloc.so" > > loadmodule "/usr/local/lib/ser/modules/registrar.so" > > loadmodule "/usr/local/lib/ser/modules/auth.so" > > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > > loadmodule "/usr/local/lib/ser/modules/uri.so" > > > > modparam("auth_db|uri_db|usrloc", "db_url", > > " mysql://ser:heslo@localhost/ser") > > > > modparam("auth_db", "calculate_ha1", 1) > > modparam("auth_db", "password_column", "password") > > > > modparam("nathelper", "natping_interval", 30) > > modparam("nathelper", "ping_nated_only", 1) > > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > > > > modparam("usrloc", "db_mode", 2) > > > > modparam("registrar", "nat_flag", 6) > > > > modparam("rr", "enable_full_lr", 1) > > > > route { > > > > # > > ----------------------------------------------------------------- > > # Sanity Check Section > > # > > ----------------------------------------------------------------- > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > break; > > }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); > > break; > > }; > > > > # > > ----------------------------------------------------------------- > > # Record Route Section > > # > > ----------------------------------------------------------------- > > if (method!="REGISTER") { > > record_route(); > > }; > > > > if (method=="BYE" || method=="CANCEL") { > > unforce_rtp_proxy(); > > } > > > > # > > ----------------------------------------------------------------- > > # Loose Route Section > > # > > ----------------------------------------------------------------- > > if (loose_route()) { > > > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > if (nat_uac_test("19")) { > > setflag(6); > > force_rport(); > > fix_nated_contact(); > > }; > > force_rtp_proxy("l"); > > }; > > route(1); > > break; > > }; > > > > # > > ----------------------------------------------------------------- > > # Call Type Processing Section > > # > > ----------------------------------------------------------------- > > > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (uri==myself) { > > > > if (method=="CANCEL") { > > route(3); > > break; > > } else if (method=="INVITE") { > > route(3); > > break; > > } else if (method=="REGISTER") { > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > }; > > > > route(1); > > } > > > > route[1] { > > > > # > > ----------------------------------------------------------------- > > # Default Message Handler > > # > > ----------------------------------------------------------------- > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > if (method=="INVITE" && isflagset(6)) { > > unforce_rtp_proxy(); > > }; > > sl_reply_error(); > > }; > > } > > > > route[2] { > > > > # > > ----------------------------------------------------------------- > > # REGISTER Message Handler > > # ---------------------------------------------------------------- > > > > if (!search("^Contact:\ +\*") && nat_uac_test("19")) { > > setflag(6); > > fix_nated_register(); > > force_rport(); > > }; > > > > sl_send_reply("100", "Trying"); > > > > if (!www_authorize("","subscriber")) { > > www_challenge("","0"); > > break; > > }; > > > > if (!check_to()) { > > sl_send_reply("401", "Unauthorized"); > > break; > > }; > > > > consume_credentials(); > > > > if (!save("location")) { > > sl_reply_error(); > > }; > > } > > > > route[3] { > > > > # > > ----------------------------------------------------------------- > > # CANCEL and INVITE Message Handler > > # > > ----------------------------------------------------------------- > > > > if (nat_uac_test("19")) { > > setflag(6); > > } > > > > lookup("aliases"); > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > > > if (isflagset(6)) { > > force_rport(); > > fix_nated_contact(); > > force_rtp_proxy(); > > }; > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > if(isflagset(6)) { > > unforce_rtp_proxy(); > > } > > sl_reply_error(); > > }; > > } > > > > onreply_route[1] { > > > > if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { > > if (!search("^Content-Length:\ +0")) { > > force_rtp_proxy(); > > }; > > }; > > > > if (nat_uac_test("1")) { > > fix_nated_contact(); > > }; > > } > > > > > > > > > > Thanks, > > > > > > Ali G. >

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[Serusers] append_rpid_hf
by Sebastian Kühner 14 Sep '05

14 Sep '05

Hello, I tried to use the append_rpid_hf command. I read in the internet that the value is saved in the subscriber table in the column rpid. Is that all I have to do? I noticed that there's no information in the header (with ngrep). Thanks! Sebastian

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RE: [Serusers] UAC module
by Pavol Segeč 14 Sep '05

14 Sep '05

For which OS platform is it compiled? ps > -----Original Message----- > From: Chris St Denis [mailto:chris@aebc.com] > Sent: Tuesday, September 13, 2005 7:18 PM > To: 'Pavol Segec'; serusers(a)lists.iptel.org > Cc: chris(a)aebc.com > Subject: RE: [Serusers] UAC module > > Somebody wrote a backport to 0.9.x. I don't remember where I > got it from, but I've attached it here. > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Pavol Segec > Sent: Tuesday, September 13, 2005 8:39 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] UAC module > > Hi, > > I see that 0.9.3 and 0.9.4 do not contain UAC module (src > package neither), just as I see last CVS snapshot have it. > How can I add UAC module for 0.9.x server? > Is there some way? > > thanks > > pavol > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] serctl for postgres
by Joao Pereira 14 Sep '05

14 Sep '05

Hi I have a version of serctl ($Id: serpgctl,v 1.1 2003/04/08 01:25:35 lgfausak Exp $) that is getting the "400 ul_add: flags expected" error. Iqbal said that maybe thats not the serctl version for my SER. I m using ser-0.9.0 and I dont have a postgres version of serctl for this version of SER. Does someone has the serctl_postgres for ser-0.9.0 ? Thanks João

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[Serusers] avpops
by Sebastian Kühner 14 Sep '05

14 Sep '05

Hello, Can anyone of you give me an example how to get data from a database with the avpops-module? I want to get a value from a special table and a special column... I didn't find any examples in the internet and the commands only don't help me very much... Many thanks! Sebastian

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[Serusers] nathelper RTP Proxy config
by Daniel Corbe 14 Sep '05

14 Sep '05

Hello, Seeing as how SER can handle a hell of alot more traffic than rtpproxy, is there a way to set up multiple rtpproxy servers and then configure SER to use them in a load balancing/failover fashion? Here's my settings for nathelper currently: # -- nathelper params -- modparam("nathelper", "natping_interval", 10) modparam("registrar", "nat_flag", 6) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "udp:10.10.17.5:22222") Thanks! -Daniel

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[Serusers] acc+mysql ser-0.9.4
by Szabolcs Szasz 14 Sep '05

14 Sep '05

Hi guys! I would like to upgrade my SER from 0.8.14 to 0.9.3 or 0.9.4 on my FC3 box. I have a problem, I can't start SER with acc module (segmentation fault). I have this problem in both 0.9.3 and 0.9.4 versions. This is the end of the debug messages (debug level:4) when I start my SER: ........... WARNING: no fork mode 0(32449) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: acc acc - initializing 0(0) find_export: found <load_tm> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <register_tmcb> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_newtran> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_relay_to_tcp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_relay_to_udp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_reply_with_body> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_is_local> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_get_trans_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_lookup_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_add_blind> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_forward_nonack> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_request_within> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_request_outside> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_request> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <new_dlg_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <dlg_response_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <new_dlg_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <dlg_request_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <free_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <print_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_gett> in module tm [/usr/local/lib/ser/modules/tm.so] Segmentation fault (core dumped)

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[Serusers] Recommendations for a fast/flexible B2BUA
by Daniel Corbe 14 Sep '05

14 Sep '05

Anyone know of a good/fast/flexible B2BUA? I'm looking to do some LCR, Billing (prepaid, postpaid, credit accounts, etc), and just some general call supervision. I see that Vovida's b2bua hasn't seen a fresh release in over 2 years now and I'm wondering if anyone still has this deployed and if it still offers the kind of flexibility that would be required to build an LCR/Billing engine. Does anyone have asterisk deployed as a B2BUA? If so, is it fast and does it provide a decent amount of flexibility? Have you had to throw a session border controller in front of it? Any other recommendations on a B2BUA? Thanks. -Daniel

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Re: [Serusers] SIP Express Router now is commercial????
by Greger V. Teigre 14 Sep '05

14 Sep '05

[new post of previous summary of Tekelec acquisition after input from Daniel-Constantin Mierla on this list earlier today. This post can also be found at http://onsip.org/ ] AFAIK, Tekelec bought iptelorg.com with its intellectual property (commercial extensions/extras/related software to SER) and its employees. Iptel.org was NOT part of the deal and is still sponsored by FOKUS Fraunhofer (as stated on the iptel.org website). SER is an independent GPL project and Tekelec cannot do anything about that. However, iptelorg.com sponsored development resources on the open source SER. Tekelec is free to decide to stop this sponsoring. I cannot see how this can be in their interest, but if the focus on the development they want to do with SER is too far away the open source project, the project may not give Tekelec enough value. If this happens, the developers must decide whether to continue in their spare time, quit SER development, or look for employment with another company willing to sponsor hours on open source SER development. In addition, SER has MANY other developers who were not involved at all with iptelorg.com. Of course, as some of the most active coders were with iptelorg.com, such a decision may impact SER's development. Also, note that the OpenSER project has branched off with many main contributions from Voice System. The Voice System developers are not affected by the Tekelec deal. However, as OpenSER since openser has a published policy to stay compatible with ser and thus continously incorporate SER code into OpenSER, OpenSER will be equally affected (in terms of reduced code contribution) if those contributing this SER code are prevented from working on the code in their day-time job. Again, both SER and OpenSER have many non-iptelorg.com contributors to the code bases and the development will not halt.

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[Serusers] Mediaproxy
by Alberto 14 Sep '05

14 Sep '05

I have a doubt about Mediaproxy. I've read, if one o more SIP clients are behind NAT the traffic RTP, all the traffic, it will cross the server (ser and mediaproxy same server). I have verified it. Is There some way that is not thus? Or it is the only form to work the Mediaproxy (always to be and mediaproxy same server). Thanks, -- Alberto

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[Serusers] SIP Express Router now is commercial????
by carlos ivan grajales 14 Sep '05

14 Sep '05

Hi Guys, And now what Tekelec bought to iptelorg that it will be of SER??? your users list??? and GPL software ??? Sincerely, CArlos Ivan Grajales M. Colombia, South America --- serusers-request(a)lists.iptel.org wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, > visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body > 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it > is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. I cannot compile "check_from()" method > although I load module > "uri" (tulay sazak) > 2. Compilation Error on Mac OS (Darwin) - SEMS.. > (Amos Nungu) > 3. call me (harry gaillac) > 4. SER + Mediaproxy sl_send_reply: I won't send a > reply for > ACK!! (GT) > 5. HELP! duplicate accounting (torres lloyd) > 6. Re: SER + Mediaproxy sl_send_reply: I won't > send a reply for > ACK!! (Greger V. Teigre) > 7. Re: Re: SER + Mediaproxy sl_send_reply: I > won't send a reply > for ACK!! (GT) > 8. Re: t_check_status in failure_route (Klaus > Darilion) > 9. Media proxy idle calls. (Vikash Khatuwala) > 10. Re: t_check_status in failure_route (Juha > Heinanen) > 11. Re: t_check_status in failure_route (Andreas > Granig) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 11 Sep 2005 04:12:08 -0700 (PDT) > From: tulay sazak <tulaysazak(a)yahoo.com> > Subject: [Serusers] I cannot compile "check_from()" > method although I > load module "uri" > To: serusers(a)lists.iptel.org > Message-ID: > <20050911111208.51148.qmail(a)web52108.mail.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > Hello everbody, > > I loaded module "uri" and my ser.cfg is as below: > > ..... > loadmodule "/usr/local/lib/ser/modules/uri.so" > ..... > if (method=="INVITE") { > if (!radius_proxy_authorize("iptel.org")) { > proxy_challenge("iptel.org", "1"); > break; > } > > if (!check_from()) { > sl_send_reply("403", "Forbidden"); > break; > } > } > ........ > > When I run ser, it says: > > 0(10457) find_export: <check_from> not found > 0(10457) find_export: <check_from> not found > 0(10457) parse error (183,35-36): unknown command, > missing loadmodule? > > Can you see what am I doing wrong ?? > > Thanks in advance > Regards > > Tulay > > > > > > > > ______________________________________________________ > > Yahoo! for Good > Watch the Hurricane Katrina Shelter From The Storm > concert > http://advision.webevents.yahoo.com/shelter > > > > ------------------------------ > > Message: 2 > Date: Sun, 11 Sep 2005 15:05:54 +0200 > From: Amos Nungu <amos.nungu(a)gmail.com> > Subject: [Serusers] Compilation Error on Mac OS > (Darwin) - SEMS.. > To: serusers(a)lists.iptel.org > Message-ID: > <86caff8805091106053263560b(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'm installing SEMS-0.9 on Mac OS (Darwin 8.2.0). > upon running the make > all command, I got an error as shown below > ------------------------------------------------- > # make all > make[1]: Nothing to be done for `deps'. > g++ -o sems AmApi.o AmAudio.o AmCmd.o AmConfig.o > AmEventQueue.o > AmIcmpWatcher.o AmMail.o AmPlugIn.o AmRequest.o > AmRtpPacket.o > AmRtpPacketTracer.o AmRtpScheduler.o AmRtpStream.o > AmSdp.o AmServer.o > AmSession.o AmSmtpClient.o AmThread.o AmUtils.o > EmailTemplate.o > SemsConfiguration.o SerClient.o SerDBQuery.o log.o > sems.o -lm -ldl > -lsocket -lnsl -lpthread > /usr/bin/ld: can't locate file for: -lsocket > collect2: ld returned 1 exit status > make[1]: *** [sems] Error 1 > make: [all] Error 2 (ignored) > ------------------------------------------------------- > > Any idea on how to go about it will be great. > > Regards > > > > ------------------------------ > > Message: 3 > Date: Sun, 11 Sep 2005 19:19:03 +0200 (CEST) > From: harry gaillac <gaillacharry(a)yahoo.fr> > Subject: [Serusers] call me > To: serusers(a)lists.iptel.org > Message-ID: > <20050911171904.17161.qmail(a)web25502.mail.ukl.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > hello, > > Please to call me at for tests. > > sip:84@nxs;yi.org > or > sip:85@nxs;yi.org > or > sip:86@nxs;yi.org > or > sip:87@nxs;yi.org > > Regards > Harry > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > ------------------------------ > > Message: 4 > Date: Sun, 11 Sep 2005 20:11:26 -0800 > From: "GT" <geotelcom(a)yahoo.fr> > Subject: [Serusers] SER + Mediaproxy sl_send_reply: > I won't send a > reply for ACK!! > To: serusers(a)lists.iptel.org <serusers(a)lists.iptel.org> > Message-ID: <20050911181312.D169319C6(a)fox.iptel.org> > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > === message truncated === __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Problem with command "serctl add"
by Alberto 14 Sep '05

14 Sep '05

Hi, After that I install SER and I run the command: serctl add 1000 12345 mil(a)domain.net I haven't any problem. But, when I restart the server and I run the command: serctl add 2000 54321 dosmil(a)domain.net I get the next problem: ERROR 1045 (28000): Access denied for user 'serro'@'localhost' (using password: YES) error: SER/FIFO not accessible: 2 I have tried the next: (but, I haven't resolved the problem) shell# mysql -p mysql> used ser; mysql> SET PASSWORD FOR 'serro'@'localhost' = PASSWORD('thepassword'); mysql> SET PASSWORD FOR 'ser'@'localhost' = PASSWORD('thepassword'); Some idea? Thanks, Alberto.

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[Serusers] SER <==> SEMS
by Pavol Segeč 14 Sep '05

14 Sep '05

Hi, Ser, sems is leaving following error message in my suyslog ep 14 12:12:32 mail /usr/sbin/ser[23536]: Warning: run_failure_handlers: no UAC support (0, 256) Sep 14 12:12:32 mail Sems[23635]: Error: (AmRequest.cpp)(bye)(256): AmRequest::b ye: 481 Call/Transaction Does Not Exist What does it means? pavol

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[Serusers] handle_subscription
by Bin Wang 14 Sep '05

14 Sep '05

hello, I'm using the methode handle_subscribtion for testing purposes. It's working, but the to-tag and from-tag in notify don't come up to our testing expectations: - to will be put in from and a tag will be generated - from will be put in to including tag but we need: the to- and from-tags in subscribe should be the from- and to-tags in notify, without generating a new tag or so - like shown above: Request-Line: SUBSCRIBE .... Message Header Via: ... ... To: sip:xyz@blabla.com; tag=b27e1a1d33761e... From: JP <sip:abc@blabla.com; tag=3c85d92779ea08e ... and notify should look like: Request-Line: NOTIFY .... Message Header via: ... ... To: JP <sip:abc@blabla.com; tag=3c85d92779ea08e From: sip:xyz@blabla.com; tag=b27e1a1d33761e... Is it possible to change the code and where do I have to change it? thanks, Bin -- GMX DSL = Maximale Leistung zum minimalen Preis! 2000 MB nur 2,99, Flatrate ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl

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[Serusers] Sip EXpress router (0.9.4 (i386/linux))
by fedora＠voice.dyndns.tv 14 Sep '05

14 Sep '05

[cycle #: 13; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.9.4 (i386/linux)) Now: Wed Sep 14 17:26:25 2005 Up Since: Wed Sep 14 16:46:13 2005 Up time: 2412 [sec] Transaction Statistics Current: 0 (0 waiting) Total: 0 (0 local) Replied localy: 0 Completion status 6xx: 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 Stateless Server Statistics 200: 97 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 35 403: 0 404: 0 407: 0 408: 0 483: 0 4xx: 0 500: 0 5xx: 0 6xx: 0 xxx: 132 failures: 0 UsrLoc Stats Domain Registered Expired 'location' 5 0 'aliases' 52 0 BYE sip:8991@221.186.150.67:5062 SIP/2.0 Via: SIP/2.0/UDP 221.186.150.68:5060;rport;branch=z9hG4bK350F835C78BE44D5AF06441318FB65AE From: 8000 <sip:8000@voice.dyndns.tv>;tag=4202623510 To: <sip:8991@voice.dyndns.tv>;tag=as4431c5d7 Contact: <sip:8000@221.186.150.68:5060> Route: <sip:221.186.150.67;ftag=4202623510;lr=on> Call-ID: 973AC72D-4AFD-4C5A-96BF-CFF6E27063CB(a)221.186.150.68 CSeq: 59653 BYE User-Agent: X-PRO build 1082 Content-Length: 0 BYE sip:8000@221.186.150.68:5060 SIP/2.0 Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK7b866277;rport Route: <sip:8000@221.186.150.68:5060> From: <sip:8991@voice.dyndns.tv>;tag=as4431c5d7 To: 8000 <sip:8000@voice.dyndns.tv>;tag=4202623510 Contact: <sip:8991@221.186.150.67:5062> Call-ID: 973AC72D-4AFD-4C5A-96BF-CFF6E27063CB(a)221.186.150.68 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0

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[Serusers] First ENUM/SIP conference in the Netherlands
by Adrian Georgescu 14 Sep '05

14 Sep '05

For those interested: http://www.isoc.nl/activ/2005-ENUMSIP/ taking place in the Hague on the 14-th of October. High profile speakers at a fraction of cost compared to other similar conferences. Same week as RIPE 51 in Amsterdam, for those around it would be nice to meet there as well. Best regards, Adrian Georgescu

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Re: [Serusers] No OK reply for REGISTER request ?
by Samuel Osorio Calvo 14 Sep '05

14 Sep '05

serctl ul show Samuel Unclassified. >>> Bhaskar Bhumkar <Bhaskar.Bhumkar(a)Sun.COM> 09/14/05 04:07AM >>> Thanks for the tip. It works - I received the 200 OK response for the REGISTER request. However, I can't see the new user name from the serweb tool (User Management screen) Is there any visual way to see the new user either by serweb or serctl? Chris St Denis wrote: > Probably the uri==myself isn't matching. > > You need to define aliases for yourself in the top of the config file. > alias="registrar.mydomain.com" > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of Bhaskar Bhumkar > Sent: Monday, September 12, 2005 6:12 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] No OK reply for REGISTER request ? > > Hi, > > I have set up a SER server and am testing it for REGISTER request. > I am sending following REGISTER request to the server : > > REGISTER sip:registrar.mydomain.com SIP/2.0 > Via: SIP/2.0/UDP bobspc.mydomain.com:5060;branch=z9hG4bKnashds7 > Max-Forwards: 70 > To: Bob <sip:bob@mydomain.com> > From: Bob <sip:bob@mydomain.com>;tag=456248 > Call-ID: 843817637684230@998sdasdh09 > CSeq: 1826 REGISTER > Contact: <sip:bob@192.0.2.4> > Expires: 7200 > Content-Length: 0 > > However, I am not getting OK reply from the server. How can I verify > user's registration at the SER? I tried to modify the ser.cfg; but > no success. Following is part of my ser.cfg : > > > ........ > if (uri==myself) { > if (method=="REGISTER") { > > > # Uncomment this if you want to use digest authentication and set > domainname > # if (!www_authorize("domainname", > "subscriber")) { > # www_challenge("domainname", "0"); > # break; > # }; > > save("location"); > break; > }; > > ........... > > > Any help would be appreciated on this. > > Thanks in advance. > > > > Regards, > Bhaskar... > > > _______________________________________________ > 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 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] new stable release: ser-0.9.4-fc1.0 SOLVED !
by fedora＠voice.dyndns.tv 14 Sep '05

14 Sep '05

+ /usr/lib/rpm/find-debuginfo.sh /usr/src/redhat/BUILD/ser-0.9.4 extracting debug info from /var/tmp/ser-0.9.4-root/usr/sbin/ser extracting debug info from /var/tmp/ser-0.9.4-root/usr/sbin/gen_ha1 extracting debug info from /var/tmp/ser-0.9.4-root/usr/sbin/serunix extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/acc.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth_db.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth_diameter.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/avp.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/avp_db.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/avpops.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/dbtext.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/dispatcher.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/diversion.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/domain.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/enum.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/exec.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/flatstore.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/gflags.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/group.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/lcr.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/mangler.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/maxfwd.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/mediaproxy.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/msilo.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/nathelper.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/options.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/pdt.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/permissions.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/pike.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/print.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/registrar.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/rr.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/sl.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/sms.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/speeddial.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/textops.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/tm.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uac.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uri.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uri_db.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/usrloc.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/xlog.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/mysql.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/jabber.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth_radius.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/group_radius.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uri_radius.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/cpl-c.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/pa.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/postgres.so -rw-r--r-- 1 root root 747155 9月 14 15:02 ser-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 1921633 9月 14 15:02 ser-0.9.4-fc1.0.src.rpm -rw-r--r-- 1 root root 1914146 9月 14 15:10 ser-0.9.4_src.tar.gz -rw-r--r-- 1 root root 70821 9月 14 15:02 ser-cpl-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 3611704 9月 14 15:03 ser-debuginfo-0.9.4-fc1.0.i38 6.rpm -rw-r--r-- 1 root root 64844 9月 14 15:02 ser-jabber-0.9.4-fc1.0.i386.r pm -rw-r--r-- 1 root root 114645 9月 14 15:02 ser-mysql-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 53330 9月 14 15:02 ser-pa-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 20923 9月 14 15:02 ser-postgres-0.9.4-fc1.0.i386 .rpm -rw-r--r-- 1 root root 21675 9月 14 15:02 ser-radius-0.9.4-fc1.0.i386.r pm -rw-r--r-- 1 root root 9356 9月 14 15:09 ser.spec.fedora-0.9.4

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[Serusers] No OK reply for REGISTER request ?
by Bhaskar Bhumkar 14 Sep '05

14 Sep '05

Hi, I have set up a SER server and am testing it for REGISTER request. I am sending following REGISTER request to the server : REGISTER sip:registrar.mydomain.com SIP/2.0 Via: SIP/2.0/UDP bobspc.mydomain.com:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: Bob <sip:bob@mydomain.com> From: Bob <sip:bob@mydomain.com>;tag=456248 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Contact: <sip:bob@192.0.2.4> Expires: 7200 Content-Length: 0 However, I am not getting OK reply from the server. How can I verify user's registration at the SER? I tried to modify the ser.cfg; but no success. Following is part of my ser.cfg : ........ if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication and set domainname # if (!www_authorize("domainname", "subscriber")) { # www_challenge("domainname", "0"); # break; # }; save("location"); break; }; ........... Any help would be appreciated on this. Thanks in advance. Regards, Bhaskar...

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[Serusers] RTPProxy "pulling my hairs"
by Ali Zaidi 14 Sep '05

14 Sep '05

Hi, I have trouble with RTPPRoxy. I downloaded a everything.GettingStarted.tar.gz package from Onsip, hello-world.3.04.cfg and auth-mysql.3.05.cfg work but when i use nat-rtpproxy.3.04.cfg i get following errors. [root@localhost root]# /etc/init.d/rtpproxy start [root@localhost root]# /etc/init.d/ser restart Stopping ser: [FAILED] Starting ser: [ OK ] [root@localhost root]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config [root@localhost root]# By the way i don't see rtpproxy.sock under \var\run\ Questions: If i use rtpptoxy, how we start it? i do "/etc/init.d/rtpproxy start" then it doesn't show if that is started or not unless i run "ps -A" Can we put SER into DMZ and do port forwarding OR SER must have Puplic IP? If we can put SER on DMZ then what other ports need to be open beside 5060? # =============================================== debug=3 fork=yes log_stderror=no listen=x.x.x.x # put your server IP address here That is MY WAN IP. port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser ") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Thanks, Ali G.

1 0

[Serusers] ser mysql database sync
by Jack Wei 14 Sep '05

14 Sep '05

i'm not familiar w/ ser enough to make the following changes, but does anyone know how to make ser update entries in table location that are present in the cache only. I have 2 ser servers and i want to use 1 database. Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Intrado 911 SER Module
by Darren Sessions 13 Sep '05

13 Sep '05

We are about to begin work on a 911 module specifically tailored to interop with Intrado's 911 service. I am curious if there is anyone out there that has either completed such a module, or might have some code already written to which we could collaborate on to get the finished product. Thanks, - Darren

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[Serusers] append_branch And New totag Problem
by Corey S. McFadden 13 Sep '05

13 Sep '05

Guys, We're working with Tekelec to solve a problem we're experiencing with one of their SIP gateway cards. (We're working with a group that hadn't heard of SER before we talked to them, so this may or may not be correct.) What happens is this: 1. Call originates from SIP PSTN GW (Tekelec unit) 2. SER routes call to UA 3. UA doesn't answer 4. Failure route happens and call is diverted to Asterisk 5. Asterisk gets the call but the SIP PSTN GW doesn't ACK the OKs and hangs up. The scenario works with a variety of UA hardware, so I didn't think anything was wrong on our side, but they're saying that on the first phase of the call the totag has one ID and then when Asterisk gets involved there's a different totag ID. (This is confirmed w/packet captures.) When the GW card gets the second totag it doesn't match a transaction and it is ignored. So, my question is... can (and should) we rewrite the totags back to the original id? Can this be done w/textops? Is this a common problem? Does this make sense? Thanks for any help, -Corey ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

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[Serusers] SER + Windows Domain
by Jaroslaw Gawron 13 Sep '05

13 Sep '05

Hi all Is there a way to integrate sip authentication with Windows domain database - to integrate function www_authorize with the Active Directory ? If anyone know how to solve this problem - any suggestions are very welcome. Best regards, Jaroslaw Gawron

3 5

[Serusers] BAD REQUEST from PSTN gateway
by Ashutosh Kumar 13 Sep '05

13 Sep '05

Hi, Until now my ser was working fine with pstn gateway and everything. Then I added session timer code, then removed it. After that calls to pstn are getting a BAD REQUEST by the gateway, as per the following ngrep output. But from X-Ten phone, the calls are fine. -- # U x.x.x.112:5060 -> x.x.x.100:5060 INVITE sip:011919899518291@x.x.x.100 SIP/2.0. Record-Route: <sip:x.x.x.112;ftag=6666f47f;lr=on>. Via: SIP/2.0/UDP x.x.x.112;branch=z9hG4bKe7d7.61cc1fd5.0. Via: SIP/2.0/UDP 127.0.0.1:5062;received=61.16.236.34;branch=z9hG4bKnp1718443690-48380229127. 0.0.1;rport=23943. From: "ashu" <sip:satya@x.x.x.112>;tag=6666f47f. To: <sip:011919899518291@x.x.x.112>. Call-ID: 1718023321-4700bac4(a)127.0.0.1. CSeq: 1 INVITE. User-Agent: Nero SIPPS IP Phone Version 2.0.51.16. Expires: 120. Accept: application/sdp. Content-Type: application/sdp. Content-Length: 271. Contact: <sip:satya@a.b.c.d:23943>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, INFO. . v=0. o=SIPPS 1718024970 1718024973 IN IP4 0.0.0.0. s=SIP call NPC0. c=IN IP4 0.0.0.0. t=0 0. m=audio 30000 RTP/AVP 0 8 97 2 3. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=fmtp:97 mode=20. a=sendrecv. # U x.x.x.100:5060 -> x.x.x.112:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP x.x.x.112;branch=z9hG4bKe7d7.61cc1fd5.0. Via: SIP/2.0/UDP 127.0.0.1:5062;received=61.16.236.34;branch=z9hG4bKnp1718443690-48380229127. 0.0.1;rport=23943. From: "ashu" <sip:satya@x.x.x.112>;tag=6666f47f. To: <sip:011919899518291@x.x.x.112>. Call-ID: 1718023321-4700bac4(a)127.0.0.1. CSeq: 1 INVITE. Content-Length: 0. . # U x.x.x.100:5060 -> x.x.x.112:5060 SIP/2.0 400 Bad Request. To: <sip:011919899518291@x.x.x.112>;tag=3335618647-158732. From: "ashu" <sip:satya@x.x.x.112>;tag=6666f47f. Call-ID: 1718023321-4700bac4(a)127.0.0.1. CSeq: 1 INVITE. Contact: sip:011919899518291@x.x.x.100:5060. Via: SIP/2.0/UDP x.x.x.112;branch=z9hG4bKe7d7.61cc1fd5.0. Via: SIP/2.0/UDP 127.0.0.1:5062;received=61.16.236.34;branch=z9hG4bKnp1718443690-48380229127. 0.0.1;rport=23943. Content-Length: 0. Regards, Ashutosh Kumar Chetu, Inc.

1 0

RE: [Users] how to modify record route header with advertised_address?
by Samuel Osorio Calvo 13 Sep '05

13 Sep '05

You can try to use record_route_preset, but be carefull!!!!! Samuel. Unclassified. >>> Gustavo García <ggb(a)tid.es> 09/13/05 05:20PM >>> Thank you for the information Bogdan. Acording to the "openser core cookbook" (http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook): "represents the address advertised in Via header and other destination lumps (e.g RR header)" May be you might want to change it. Best regards. G. -----Mensaje original----- De: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Enviado el: martes, 13 de septiembre de 2005 17:13 Para: Gustavo García CC: 'Klaus Darilion'; users(a)openser.org Asunto: Re: [Users] how to modify record route header with advertised_address ? Hi Gustavo, the RR header will be filled with IP address only on send time - the IP of the outgoing interface will be sent all the time. The "advertised_address" is not used in this case (as for via). regards, Bogdan Gustavo García wrote: >Thank you Klaus, I was already using that function, but I wanted to know why >doesn't work the advertised_address property with record route. Is a bug in >my script, a bug in openser or is this the expected working? > >G. > >-----Mensaje original----- >De: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >Enviado el: martes, 13 de septiembre de 2005 10:07 >Para: Gustavo García >CC: users(a)openser.org >Asunto: Re: [Users] how to modify record route header with >advertised_address ? > >a workaround: http://openser.org/docs/modules/0.10.x/rr.html#AEN163 > >regards >klaus > >Gustavo García wrote: > > >>Hi, >> >> >> >>I've configured an address with advertised_address. >> >>I can see that it's used in Via address but not in record route. What >>could be the reason? >> >> >> >>Thanks a lot >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] cone and symmectric nat
by Rick Thompson 13 Sep '05

13 Sep '05

Can anyone explain to me why if I use fix_nated_sdp on symmetric nat'd ua's they work but they don't work if I use force_rtp_proxy (one way audio). Yet if I use force_rtp_proxy on cone nated ua's they work but don't work on fix_nated_sdp? Richard C. Thompson

1 0

[Serusers] UAC module
by Pavol Segec 13 Sep '05

13 Sep '05

Hi, I see that 0.9.3 and 0.9.4 do not contain UAC module (src package neither), just as I see last CVS snapshot have it. How can I add UAC module for 0.9.x server? Is there some way? thanks pavol

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Re: [Serusers] NEED HELP!!!
by harry gaillac 13 Sep '05

13 Sep '05

Hello Alberto, Translate please ! Harry --- Alberto <alberto.ipt(a)telefonica.net> a écrit : > Francés tenias que ser. > > ----- Original Message ----- > From: "harry gaillac" <gaillacharry(a)yahoo.fr> > To: <serusers(a)lists.iptel.org> > Sent: Tuesday, September 13, 2005 2:51 PM > Subject: [Serusers] NEED HELP!!! > > > > Hello ser users, > > > > You needn't to reply to my ser configuration > problems. > > I solve them myself . > > > > Kinds regards > > > > Harry > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Users] TCP QUESTION
by Shiran Guez 13 Sep '05

13 Sep '05

Hi I am new in openser and I have installed openser stable release. I have tested the SER with Xten and Linksys and it works fine, then I wanted to try to do TCP calls so I have configured MSN Messenger to work with TCP and tried to register but got rejected I have attached the cfg file I am working with, can some one help? Shiran

2 1

[Serusers] shmem errors while running ser 0.9.4
by Hilco de Lathouder 13 Sep '05

13 Sep '05

Im running ser 0.9.4 on a Fedora Core 3 box. Im using mysql auth & storage. If a client wants to register, the debug information displays the following (below): What could be the cause of this problem ? TIA Hilco ====================================== textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 1(0) INFO: fifo process starting: 19593 1(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 1(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 11(19623) ERROR: build_req_buf_from_sip_req: out of memory 11(19623) ERROR: print_uac_request: no pkg_mem 11(19623) ERROR: t_forward_nonack: failure to add branches 16(19638) ERROR: print_uac_request: no shmem 16(19638) ERROR: t_forward_nonack: failure to add branches 17(19641) ERROR: relay_reply: cannot alloc reply shmem 17(19641) ERROR: _reply_light: cannot allocate shmem buffer ==========================================

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[Serusers] Forwarding to SIP proxy
by Marc Khayat 13 Sep '05

13 Sep '05

If I do the following: GW calls SIP proxy : works However, I'm trying to forward (stateful) a call to a SIP Proxy, but it seems that it's failing. This is my scenario: GW -> SER calls SIP proxy: fails. This is my ser.cfg if (uri=~"^sip:123") { t_relay_to_udp("Proxy_IP", "5060"); break; }; I thought that it's the same as a GW, but it seems that I'm wrong. Help is appreciated :)

1 0

[Serusers] SER lock fails
by Andreas Granig 13 Sep '05

13 Sep '05

Hi, every now and then my SERs (0.9.4-rc3) hang with a load of 12 or so (I think it's because 12 SER children are running). I attached with gdb but have no backtrace anymore, however I know it was the LOCK_REPLIES( t ); in modules/tm/t_reply.c:860 It seems it can't get the lock and thus loops in get_lock(...) in fastlock.h:160. I can't reproduce this and it happens just ones a week or every other week... Any ideas? Any

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[Serusers] new stable release: ser 0.9.4
by Andrei Pelinescu-Onciul 13 Sep '05

13 Sep '05

ser 0.9.4 is the new stable release. This is a "bug fix" release, it contains no new features, just bug fixes. Upgrading from 0.9.3 is very easy: just replace your 0.9.3 ser with the new version (the 0.9.3 config should work without any changes and the same goes for sems, serweb and the mysql tables). 0.9.4 uses the same cvs branch as 0.9.3: rel_0_9_0. The new release can be downloaded from: ftp://ftp.berlios.de/pub/ser/0.9.4/ (see ftp://ftp.berlios.de/pub/ser/0.9.4/README for more information). If you use debian you can user the apt.sip-router.org ser debian repository by adding the following lines: deb http://apt.sip-router.org/debian stable main contrib non-free deb http://apt.sip-router.org/debian testing main contrib non-free deb http://apt.sip-router.org/debian unstable main contrib non-free to your /etc/apt/sources.list and then apt-get install ser ser-cpl-module ser-jabber-module ser-mysq-module ser-pa-module ser-postgres-module ser-radius-modules to install all the ser packages. Changes: general: - gcc 4.0 support - mutlicast options are set for all the sockets - mediaproxy: memory leak, unchecked memory allocations - postgress: some bugs and cleanups (compiles cleanly now) - tm: shm cloned lumps (SER-55) - tm: t_fifo: 64-bit fix - tm: dst_uri clone fix - tm: t_newtran/t_restransmit* races (!) - usloc: memory leak solaris: - serctl, better ISA detection, signals *bsd: - multicast options are properly set openbsd: - tcp 64 bit fix Andrei

1 0

[Serusers] NEED HELP!!!
by harry gaillac 13 Sep '05

13 Sep '05

Hello ser users, You needn't to reply to my ser configuration problems. I solve them myself . Kinds regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

RE: [Serusers] UAC module
by Pavol Segeč 13 Sep '05

13 Sep '05

Him Thanks, I think and I found that I don't have uac.to module in my /usr/lib/ser/modules at all, how can I add a new module to my installed ser package? thanks pavol > -----Original Message----- > From: Steve Blair [mailto:blairs@isc.upenn.edu] > Sent: Tuesday, September 13, 2005 2:30 PM > To: Pavol Segeč > Subject: Re: [Serusers] UAC module > > > Look at or around line 641 for the offending command. You can > check the source modules directory to make sure that command > and it's load module are actually loaded. The other thing to > check is punctuation. Sometimes a missing ")" or ";" will > through off the command parser and give error messages for > lines where and error doesn't exist. > > _Steve > > Pavol Segeč wrote: > > >Hi, > > > >I'd like use uac_replace_from("sip:batman@gotham.org") > function of the > >UAC module. UAS needs loaded tm.so, I have it, but ser -c is > reporting error: > > > > (11552) parse error (641,21-22): unknown command, > missing loadmodule? > > > >Some advice? > > > >Thanks, > > > >pavol > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > -- > > ISC Network Engineering > The University of Pennsylvania > 3401 Walnut Street, Suite 221A > Philadelphia, PA 19104 > > > voice: 215-573-8396 > > 215-746-8001 > > fax: 215-898-9348 > > sip:blairs@upenn.edu >

1 0

[Serusers] sip phone system can connect with public telephone network ?
by crazy 13 Sep '05

13 Sep '05

Hi all If I want sip phone can phone to the public telephone network, what can I do for it? or is there no way to achieve it?

2 1

[Serusers] UAC module
by Pavol Segeč 13 Sep '05

13 Sep '05

Hi, I'd like use uac_replace_from("sip:batman@gotham.org") function of the UAC module. UAS needs loaded tm.so, I have it, but ser -c is reporting error: (11552) parse error (641,21-22): unknown command, missing loadmodule? Some advice? Thanks, pavol

1 0

[Serusers] ser script language
by Rick Thompson 13 Sep '05

13 Sep '05

Hi All I've search the whole world over and I can't seem to find a good resource for the ser scripting language that explains the syntax, operators and structure of ser commands. I found lots on the modules but not on the main ser.cfg. Can anyone give me a specific link to a doc? Thanks Rick

3 2

Re: [Serusers] ser project
by harry gaillac 13 Sep '05

13 Sep '05

Dear Cesc, Is it easier to be amazed than helping people ? Harry --- Cesc <cesc.santa(a)gmail.com> a écrit : > Hi harry, > i can't wait to see what prank you pull with your > next email ... > not that i care, but at least i get a good laugh. > Sincerely amazed, > Cesc > > On 9/8/05, harry gaillac <gaillacharry(a)yahoo.fr> > wrote: > > > > Hello, > > > > Forget this mail ! > > > http://lists.iptel.org/pipermail/serusers/2005-September/023710.html > > > > Harry > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

6 7

[Serusers] NEED HELP !!!
by harry gaillac 13 Sep '05

13 Sep '05

Hello, I've been spending much time to understand why SER don't reply to caller with public ip in SDP message. I try both rtpproxy and mediaproxy. when caller/callee are both registered in registrar server they are no problem. I can't solve it !!!! Harry My ser.cfg # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 Exp $ # # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@serveur1.home.net/ser" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # listen=80.119.5.100 # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "rtpproxy_disable", 1) # mediaproxy params -- modparam("mediaproxy", "natping_interval", 10) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # -- registrar params -- modparam("registrar", "nat_flag", 4) modparam("registrar", "use_domain", 1) # -- tm params -- modparam("tm", "fr_inv_timer", 15) modparam("tm", "fr_timer", 10) # --domain params -- modparam("domain", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("domain", "domain_table", "domain") modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # -- acc params -- modparam("acc", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("acc", "log_fmt", "miocfsp") modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -- msilo params -- modparam("msilo", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("msilo", "check_time", 10) # -- avpos params -- modparam("avpops","avp_url","mysql://ser:heslo@serveur1.home.net/ser") modparam("avpops","avp_table","usr_preferences") # ------------------------- 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(); }; if (method=="CANCEL" || method=="BYE") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; use_media_proxy(); }; route(1); break; }; # if the request is for our domains use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (is_uri_host_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="MESSAGE") { route(7); break; } else if (method=="SUBSCRIBE" || method=="NOTIFY") { route(8); break; }; }; # Requests to others domains if (!is_uri_host_local()) { # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Message Handler to others domains # if (method=="MESSAGE") { route(1); break; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!is_from_local()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); break; }; m_dump(); } route[3] { ############################# # CANCEL and INVITE handler # ############################# # SIP destinations to PSTN if (uri=~"^sip:[0-9]{4}@.*$" || uri=~"^sip:[0-9]{10}@.*$" || uri=~"^sip:06[0-9]{8}@.*$" || uri=~"^sip:00.*@.*$") { route(4); break; }; # SIP destinations to IPBX if (uri=~"^sip:8[0-3]@.*$") { route(5); break; } # Check AVPs for Voicemail if ((avp_db_load("$ruri","s:fw_voicemail/usr_preferences") && avp_check("fw_voicemail", "eq/i:1"))) { route(6); route(1); break; }; # accounting db flags setflag(2); setflag(3); acc_db_request("404 missed call", "missed_calls"); # Check Nated User Agents if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Check ACL for Voicemail if (is_user_in("Request-URI","voicemail")) { setflag(5); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (isflagset(5)) { route(6); route(1); break; } else { sl_send_reply("404", "User Not Found"); break; }; }; t_on_failure("1"); route(1); break; } route[4] { ########################## # Outgoing PSTN Handler # ########################## # Checking Authorization to PSTN if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:[0-9]{4}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:[0-9]{10}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:06[0-9]{8}@.*$") { if (!is_user_in("From", "cellphone")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:00.@.*$") { if (!is_user_in("From", "int")) { sl_send_reply("403", "Forbidden"); break; }; }; prefix("0"); rewritehostport("asterisk.home.net:5050"); route(1); break; } route[5] { ########################## # Asterisk Handler # ########################## # Checking Authorization to Asterisk if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:8[0-3]@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; }; rewritehostport("asterisk.home.net:5050"); route(1); break; } route[6] { ########################## # Voicemail Handler # ########################## revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); break; } route[7] { ########################## # Message Handler # ########################### # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); } route[8] { ################################ # SUBSCRIBE/NOTIFY Handler # ################################ # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } onreply_route[1] { ########################## # Statefull Handler # ########################## if (status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { use_media_proxy(); }; }; } failure_route[1] { ########################## # Voicemail Handler # ########################## if (t_check_status("486")) { revert_uri(); # resend to voicemail with original request URI prefix("b"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; if (t_check_status("408")) { revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; } ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Voip and Firewalls
by harry gaillac 13 Sep '05

13 Sep '05

Hello, What are the policies to apply in order to secure a network with Voip context? Which ports must be openned for signalling and which ports must be openned for rtp traffic incoming/outgoing ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 3

[Serusers] ser mysql database sync
by Jack Wei 13 Sep '05

13 Sep '05

i'm not familiar w/ ser enough to make the following changes, but does anyone know how to make ser update entries in table location that are present in the cache only. I have 2 ser servers and i want to use 1 database. Jack Wei __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

3 2

[Serusers] one way voice on public IP...ngrep
by Iqbal 13 Sep '05

13 Sep '05

Hi All the IP adds here are public, but I get one way voice, i.e from UA to pstn I can get voice, but not from pstn to UA box UA ---SER---PSTN a.b.c.d - UA x.x.x.x = SER y.y.y.y = pstn gw a.a.a.a = pstn gw 2 The c= and all seem correct, and natting problems shouldn't come into this...all ports are open, there is a FW, but that doesnt show anything, any thoughts on howto debug this Iqbal -------- U 2005/09/12 14:20:26.349238 a.b.c.d:5060 -> x.x.x.x:5060 INVITE sip:0123456789@sip.domain.com SIP/2.0. Content-Type: application/sdp. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. ## U 2005/09/12 14:20:26.393835 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Content-Length: 0. Warning: 392 x.x.x.x:5060 "Noisy feedback tells: pid=31505 req_src_ip=a.b.c.d req_src_port=5060 in_uri=sip:0123456789@sip.domain.com out_uri=sip:44123456789@y.y.y.y:5060 via_cnt==1". . # U 2005/09/12 14:20:26.394104 x.x.x.x:5060 -> y.y.y.y:5060 INVITE sip:44123456789@y.y.y.y:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Type: application/sdp. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. ## U 2005/09/12 14:20:26.401238 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Sip EXpress router (0.8.99-dev (i386/linux)). Content-Length: 0. . ###### U 2005/09/12 14:20:30.161123 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:20:30.163775 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. ## U 2005/09/12 14:20:31.368083 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:20:31.370673 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:20:31.392583 a.b.c.d:5060 -> x.x.x.x:5060 ACK sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on SIP/2.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0. . # U 2005/09/12 14:20:31.394335 x.x.x.x:5060 -> y.y.y.y:5060 ACK sip:0044123456789@a.a.a.a:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Via: SIP/2.0/UDP x.x.x.x;branch=0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0. . ############################################################################################################################################################################################################################################################################################################################### U 2005/09/12 14:22:25.033315 a.b.c.d:5060 -> x.x.x.x:5060 INVITE sip:0123456789@sip.domain.com SIP/2.0. Content-Type: application/sdp. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:25.067537 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Content-Length: 0. Warning: 392 x.x.x.x:5060 "Noisy feedback tells: pid=31508 req_src_ip=a.b.c.d req_src_port=5060 in_uri=sip:0123456789@sip.domain.com out_uri=sip:441560485332@y.y.y.y:5060 via_cnt==1". . # U 2005/09/12 14:22:25.068063 x.x.x.x:5060 -> y.y.y.y:5060 INVITE sip:441560485332@y.y.y.y:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Type: application/sdp. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:25.078167 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Sip EXpress router (0.8.99-dev (i386/linux)). Content-Length: 0. . ############ U 2005/09/12 14:22:28.963663 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:28.966645 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:30.228227 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:30.230960 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:30.253088 a.b.c.d:5060 -> x.x.x.x:5060 ACK sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on SIP/2.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0. . # U 2005/09/12 14:22:30.254767 x.x.x.x:5060 -> y.y.y.y:5060 ACK sip:0044123456789@a.a.a.a:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Via: SIP/2.0/UDP x.x.x.x;branch=0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0.

2 1

[Serusers] help!how to make ser+rtpproxy can only realize nat traversal?
by 皮面 13 Sep '05

13 Sep '05

Hi,list I have found a configure file which can make ser+rtpproxy to support NAT traversal.But I don't want to use mysql and auth or some other modules which are not concerned with nathelper.In another words,I just want to use nathelper module to realize NAT traversal as a full proxy.I did some changes on this configure file now it can be registered but still cannot realize transaction between two client:( Could sombody help me or tell me which parameter should be change! thankes! http://mianpi.blogchina.com/inc/nat-rtpproxy.3.05.cfg --------------------------------- DO YOU YAHOO!? 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒

1 0

[Serusers] Looking for a way to register many numbers with just one Register Method
by Rodrigue ESTEVE 13 Sep '05

13 Sep '05

Hi all, I'd like to know if SER offers a way to register many phone numbers with just one REGISTER request. For example I've an IP gateway connected to the PBX and to the PSTN (for backup): Phones-----> PBX--------> IP GW --------> PSTN Backup | | IP Backbone ------> SER This PBX has a lot of phone numbers associated and a main phone number (receptionist for example). I would like to make this phone numbers registered on the proxy when the IP GW is running fine. But if the PBX has 100 numbers, the IP GW has to send 100 register requests to SER, moreover I'd like to register every 5-6 minutes so that the IP GW can use PSTN backup in case of failover of the WAN link. The problem is that I'm not sure that the IP GW will scale very well for a huge quantity of phone numbers (some customers have 700 phones numbers), and all these requests will consume bandwidth. For the moment I've done a Perl Script running each time an IP GW register one number (the receptionist number for example) using the exec_dset function (exec_dset("/root/sip_registration.pl $SIP_HF_TO $SIP_SRCIP");). This perl script looks into a database all the numbers related to the "receptionist" number and sends as many register information to the SER FIFO. But I'm not very satisfied because I've to create (not yet implemented) as many temporary FIFO receipt as I receive "receptionist" register to maintain the relationship between a register request and the FIFO output (SER FIFO receipt). Moreover it's not easy to be sure that all the information passed to the FIFO have been correctly used (I'm for sure not a good programmer). So I would like to know if there is a way in SER to do this (perhaps AVP ???) so I don't have to program a crappy perl script (see below for a first release of this perl script, if somebody know how I can create a temporary unix fifo in perl without using the system command: mkifo). Thanks, Rodrigue, Here's my script: #!/usr/bin/perl -w \ use DBI; use strict; use Fcntl; #Database arguments my $db = 'DBI:mysql:telephone:localhost'; my $username = 'root'; my $passwd = 'mo1000$$'; my $table = 'association'; #Error Log my $error_log = "/var/log/sip_registration.log"; #SER FIFO arguments my $fifo_method = "ul_add"; my $fifo_receipt = "ser_receipt_fifo"; my $fifo_table = "location"; my $fifo_expiration_timer = 60; my $fifo_qvalue = 0; my $fifo_replicate = 0; my $fifo_flags = 0; #Get the uri and ip to register my $uri = $ARGV[0]; my $ip = $ARGV[1]; #Get the phone number from the uri my ($phone_number)= $uri =~ /sip:(.*)@/; #Retrieve from database the numbers associated to the uri my $dbh = DBI->connect($db, $username, $passwd); my $sql = qq{ SELECT associated FROM $table where initnumber='$phone_number' }; my $sth = $dbh->prepare($sql); $sth->execute; #Bulk registration using FIFO open(my $wr,">", "/tmp/ser_fifo"); select((select($wr), $|=1)[0]); sysopen(my $read, "/tmp/$fifo_receipt", O_RDONLY | O_NONBLOCK); my $i=1; my $associated_phone; $sth->bind_col(1,\$associated_phone); while($sth->fetch()) { if ($associated_phone != $phone_number) { print $wr ":$fifo_method:$fifo_receipt\n$fifo_table\n$associated_phone\nsip:$associate d_phone\@$ip\n$fifo_expiration_timer\\ n$fifo_qvalue\n$fifo_replicate\n$fifo_flags\n" or die; my @read_fifo; while(<$read>){ $read_fifo[0]=$_; #print "$i $read_fifo[0]"; } $i++; } } #print "ASSOCIATED PHONE : $associated_phone"; # Wait till the loop see the last associated phone, the receipt FIFO has to #be completely synchronised between Input (SER FIFO) and Ouput (Temporary #receipt FIFO) , not sure that SER works in this way. while(<$read> !~ /sip:$associated_phone\@$ip/){ #DO NOTHING } close($read); close($wr); $sth->finish(); $dbh->disconnect(); #Return uri to STDOUT to be compliant with SER exec_dset function print "$uri";

8 12

[Serusers] How to account for call forward?
by Chris St Denis 13 Sep '05

13 Sep '05

When I user has call forwarding enabled, I want them to be able to forward to a long distance number - at their own expense. Because of this, I want to account call forwarded numbers as originating as the callee and going to the destination long distance number. I would like to log this for local calls as well. How can I make this happen?

1 0

[Serusers] Module creation
by sendman 12 Sep '05

12 Sep '05

Hi folks! I'm writing a new module for SER, which wil do routing task; well now I have a little problem. The module is do routing very well, using t_relay_to_udp function, but I don't now how to setup t_on_failure route directly inside the module... So is possible do something like: cmd_function t_on_failure_f; ... if (!(t_on_failure_f = find_export("t_on_failure",1,0))) { LOG(L_NOTICE, "mod_init(): This module requires tm module\n"); return -5; } ... t_on_failure_f(msg,failure); t_relay_to_udp_f(msg,gw,foo)); Best regards. -fs

1 0

[Serusers] call tests
by harry gaillac 12 Sep '05

12 Sep '05

Please, Call sip:84@nxs.yi.org or 85 86 87 to debug my config file. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] ebay to pay $4 billion for skype [OT]
by Iqbal 12 Sep '05

12 Sep '05

well I guess there's hope for all the VOIP companies/entrepreneurs on this list...if anyone makes a billion, feel free to make donation, I accept all kinds of currencies :-) good luck Iqbal

2 2

[Serusers] Issue 5.0 of GettingStarted Now Available....
by Simon 12 Sep '05

12 Sep '05

Dear List, Issue 5.0 of the successful GettingStarted document is now available for download from http://www.onsip.org/modules/mydownloads/viewcat.php?op= <http://www.onsip.org/modules/mydownloads/viewcat.php?op=&cid=13> &cid=13 This version builds upon previous versions and as well as fixing a few bugs in the configs we have added support for call forwarding. So if you are still struggling with NAT'ed devices, or PSTN gateways, or being challenged by your Customers to add more call routing functions then this new version of GettingStarted will help. We hope you like and enjoy, as always feedback is warmly welcomed. Paul, Simon, Greger www.onsip.org <http://www.onsip.org/>

2 1

[Serusers] Re: Re: SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!!
by GT 12 Sep '05

12 Sep '05

Thanks G I'll verify all that, as a matter of fact I am internal network(s) not public but beside a FW. In MediaProxy install, it is said "must" IP public. Do I need a distinctive public IP ? Kind regards, Gtel The config will not mark non-nated devices for proxying ('serctl ul show' will not show the flag set). I suggest you verifiy that none of the EUCs are behind NAT, it should then work regardless of mediaproxy, then ensure that mediaproxy is running properly and that sessions are created (see /var/log/messages). Finally, many people fall in the trap of an iptables firewall or firewall in front of ser. g-) ---- Original Message ---- From: GT To: serusers(a)lists.iptel.org Cc: greger(a)teigre.com Sent: Monday, September 12, 2005 06:11 AM Subject: SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!! > Hello, > I am trying SER with Mediaproxy 1.2.1 as per Onsip sample config. > I am running SER & Mediaproxy on the same host rtp port range default > (35000:65000) but even for non-nated devices I get no audio until I > stop completely mediaproxy > Debugging but thinking there might be something wrong with this > config, any hint would be much appreciated, > Thanks, > Gtel

1 0

[Serusers] Can' t receive calls from other domains
by harry gaillac 12 Sep '05

12 Sep '05

Hello, I use ser-0.9.3 and mediaproxy 1.4.2 I can dial sip addresses to other domains however i can't receive calls. I 've been spending time to fix this problem without success. In fact ser don't send public ip in sdp of the callee. Thanks for help Harry $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 Exp $ # # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@serveur1.home.net/ser" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.6.86 # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "rtpproxy_disable", 1) # mediaproxy params -- modparam("mediaproxy", "natping_interval", 10) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # -- registrar params -- modparam("registrar", "nat_flag", 4) modparam("registrar", "use_domain", 1) # -- tm params -- modparam("tm", "fr_inv_timer", 15) modparam("tm", "fr_timer", 10) # --domain params -- modparam("domain", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("domain", "domain_table", "domain") modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # -- acc params -- modparam("acc", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("acc", "log_fmt", "miocfsp") modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -- msilo params -- modparam("msilo", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("msilo", "check_time", 10) # -- avpos params -- modparam("avpops","avp_url","mysql://ser:heslo@serveur1.home.net/ser") modparam("avpops","avp_table","usr_preferences") # ------------------------- 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(); }; if (method=="CANCEL" || method=="BYE") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; use_media_proxy(); }; route(1); break; }; # if the request is for our domains use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (is_uri_host_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="MESSAGE") { route(7); break; } else if (method=="SUBSCRIBE" || method=="NOTIFY") { route(8); break; }; route(1); break; }; # Requests to others domains if (!is_uri_host_local()) { # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Message Handler to others domains # if (method=="MESSAGE") { route(1); break; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!is_from_local()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); break; }; m_dump(); } route[3] { ############################# # CANCEL and INVITE handler # ############################# # SIP destinations to PSTN if (uri=~"^sip:[0-9]{4}@.*$" || uri=~"^sip:[0-9]{10}@.*$" || uri=~"^sip:06[0-9]{8}@.*$" || uri=~"^sip:00.*@.*$") { route(4); route(1); break; }; # SIP destinations to IPBX if (uri=~"^sip:8[0-3]@.*$") { route(5); route(1); break; } # Check AVPs for Voicemail if ((avp_db_load("$ruri","s:fw_voicemail/usr_preferences") && avp_check("fw_voicemail", "eq/i:1"))) { route(6); route(1); break; }; # accounting db flags setflag(2); setflag(3); acc_db_request("404 missed call", "missed_calls"); # Check Nated User Agents if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Check ACL for Voicemail if (is_user_in("Request-URI","voicemail")) { setflag(5); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (isflagset(5)) { route(6); route(1); break; } else { sl_send_reply("404", "User Not Found"); break; }; }; t_on_failure("1"); route(1); break; } route[4] { ########################## # Outgoing PSTN Handler # ########################## # Checking Authorization to PSTN if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:[0-9]{4}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:[0-9]{10}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:06[0-9]{8}@.*$") { if (!is_user_in("From", "cellphone")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:00.@.*$") { if (!is_user_in("From", "int")) { sl_send_reply("403", "Forbidden"); break; }; }; prefix("0"); rewritehostport("asterisk.home.net:5050"); break; } route[5] { ########################## # Asterisk Handler # ########################## # Checking Authorization to Asterisk if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:8[0-3]@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; }; rewritehostport("asterisk.home.net:5050"); break; } route[6] { ########################## # Voicemail Handler # ########################## revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); break; } route[7] { ########################## # Message Handler # ########################### # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); } route[8] { ################################ # SUBSCRIBE/NOTIFY Handler # ################################ # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } onreply_route[1] { ########################## # Statefull Handler # ########################## if (isflagset(4) || isflagset(6) && status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_contact(); }; } failure_route[1] { ########################## # Voicemail Handler # ########################## if (t_check_status("486")) { revert_uri(); # resend to voicemail with original request URI prefix("b"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; if (t_check_status("408")) { revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; } ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] Failure Route
by Sebastian Kühner 12 Sep '05

12 Sep '05

Hello, I have two different failure routes. One of them is for internal calls (function: mailbox on timeout or busy) and the other one is for PSTN calls (function: failover). If a user makes a call to a Virtual Number of another client (MyUser1 --> PSTN --> MyUser2), the failure routes are "mixtured". Example: Call from 44441 to 001234567890 --> Gateway --> Gateway forwards it to 44442 After a couple of seconds, the failure_route 1 (for internal calls) goes into the failure routing with the number 001234567890... but obvially there is no client with the number 001234567890. Any hints? Sebastian

1 0

FW: [Fwd: Re: [Serusers] ul_add flag and serctl]
by Aisling 12 Sep '05

12 Sep '05

Also, the following errors appeared on the shell where I started ser ERROR: parse_sip_msg_uri: bad uri aisling(a)x.x.x.x WARNING: do_action:error in expression ERROR: parse_uri: bad uri, state 0 parsed: <aisl> (4) / aisling(a)x.x.x.x (22) ERROR: parse_sip_msg_uri: bad uri aisling(a)x.x.x.x WARNING: do_action:error in expression Warning: sl_send_reply: I won't send a reply for ACK!! -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 24 August 2005 13:44 To: 'serusers(a)lists.iptel.org' Cc: 'Iqbal' Subject: RE: [Fwd: Re: [Serusers] ul_add flag and serctl] Hi Iqbal, Jan, I changed my log to /tmp/fifo in my ser.cfg and the serctl.dat file. I then tried adding an alias: Serctl alias add 7890 sip:aisling@x.x.x.x Again I got the 404 flags expected error and I checked the /tmp/fifo file. It was empty. I then did /opt/ser/sbin/serctl fifo ul_add aliases aisling(a)x.x.x.x 7890 0 1.00 0 128. (It would work unless I added those 4 parameters) It said 200 Added to table and the following appeared in my new log file: :ul_add:ser_receiver_28642 aliases aisling(a)x.x.x.x 7890 However while this said added above. When I dialed 7890 to reach the aisling(a)x.x.x.x client, a 404 was sent to my phone. Many Thanks, Aisling. -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: 24 August 2005 12:29 To: Aisling O'Driscoll Subject: [Fwd: Re: [Serusers] ul_add flag and serctl] could try this -------------------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.

3 2

[Users] Question regarding redundancy/load balancing
by Dave 12 Sep '05

12 Sep '05

Hello all: I would like to install two OpenSer servers, two database servers and a few mediaproxies. I want to achieve some level of redundancy. I have the following thoughts: 1) user DNS SRV records and this way I can do some load balancing. But my question is once a UA registers with one OpenSer server, how can I ensure that all future requests for SIP signalling will go the the same OpenSer server? What if the UA again does a DNS query before sending an INVITE and goes to a different OpenSer Server? Also how can I ensure that incoming INVITE requests will come to the UA from the "other" OpenSer server? 2) How do I handle the two database servers? If each DB server is connected to one OpenSer server, how do I ensure that what gets written to one DB is also written to the second database? Or is there some other way? I did go thru old postings on SER/OpenSER but although a lot has been discussed, no clear solution has been proposed. Any other pointers or thoughts will be highly appreciated. Dave --------------------------------- Find your next car at Yahoo! Canada Autos

2 1

[Serusers] t_check_status in failure_route
by Andreas Granig 12 Sep '05

12 Sep '05

Hi, Lately there were some discussions about t_check_status in failure_route when cancelling, where Jan (IIRC) made clear that only the lowest reply-code can be matched with t_check_status. Now I have a related problem when routing to PSTN: I use lcr-module in failure-route for routing to the next GW when 408/500/503 is received. I have noticed that I sometimes get 408 signalled from the GW (a PSTN code is mapped to it). So on one GW I mapped 408 to 480 for testing purposes to skip the fallback, the others still return 408. Now if one of the not-remapped GWs returns 408, there's the fallback to another GW. If it's the remapped one, it returns 480, but t_check_status still matches 408. I know that this special issue will be solved when I configure all my GWs to map the 408 to 480, but the question arises if this situation could come up again in other circumstances? Because if the first GW fails with 408 triggered by fr_timer, other return codes >408 from subsequent GWs can't be detected and passed back to the calling UAC without another PSTN fallback? There was a discussion about introducing a method which checks if the call is cancelled to detect 487 (don't know anymore if on the openser-lists or here), but what about other codes? Comments? Andy

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[Serusers] Media proxy idle calls.
by Vikash Khatuwala 12 Sep '05

12 Sep '05

Hi, I have setup SER 0.9.4 with Mysql and media proxy by following the instructions from the onsip getting started guide and the config file from there too. I can manage to setup the accounts and also use a hard phone (PA1688 based) and also xten to login and place calls. However I get a few problems. When I disconnect, it takes xten about 5-10seconds before it can hang up and during this time it says hanging up. Secondly after it hangs up, I can still see the call as idle status when I run sessions.py This will go away after 1min, and sometimes if I try to call back immediately after hang up it doesnt work, I have to wait for the idle call to time out. Both phones are behind NAT and so the RTP has to go through mediaproxy. When the call is active the quality is good. I havnt made any changes to ser.cfg from onsip.org other then the realm in the www_authorize and proxy_authorize. Any advise will be appreciated. Thanks and Regards, Vikash.

1 0

[Users] pa module
by Helge Waastad 12 Sep '05

12 Sep '05

Hi, Does anyone know if the pa module is going to be updated according to the changes in the SER cvs tree? Br, Hw

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[Users] RTP Relaying
by Ozan Blotter 12 Sep '05

12 Sep '05

Hi List, Case : 1. Have a database , it include SIP no's, usernames, names & surnames... 2. Register myself as admin and load a voice message in wav format... 3. Need to deliver that voice message to everyone one by one via database... 4. If callee is busy then try it a few minutes later again... 5. Close calls... Does someone interested something like this by using RTP Relaying ? If yes, please tell me your ideas.. Thanks, Ozan

2 1

[Serusers] Re: SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!!
by Greger V. Teigre 12 Sep '05

12 Sep '05

The config will not mark non-nated devices for proxying ('serctl ul show' will not show the flag set). I suggest you verifiy that none of the EUCs are behind NAT, it should then work regardless of mediaproxy, then ensure that mediaproxy is running properly and that sessions are created (see /var/log/messages). Finally, many people fall in the trap of an iptables firewall or firewall in front of ser. g-) ---- Original Message ---- From: GT To: serusers(a)lists.iptel.org Cc: greger(a)teigre.com Sent: Monday, September 12, 2005 06:11 AM Subject: SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!! > Hello, > I am trying SER with Mediaproxy 1.2.1 as per Onsip sample config. > I am running SER & Mediaproxy on the same host rtp port range default > (35000:65000) but even for non-nated devices I get no audio until I > stop completely mediaproxy > Debugging but thinking there might be something wrong with this > config, any hint would be much appreciated, > Thanks, > Gtel

1 0

[Serusers] HELP! duplicate accounting
by torres lloyd 12 Sep '05

12 Sep '05

I was able to see the accounting Logs on my radacct directory the problem was I am having duplicate START and single STOP. I also see this from my radius log file. rlm_radutmp: Login entry for NAS localhost port 5060 duplicate below is the log from a single test call. Lloyd Mon Sep 12 12:13:44 2005 Acct-Status-Type = Start <---- 1st Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 1 User-Name = "8437310(a)202.57.32.8" Calling-Station-Id = "sip:8437310@202.57.32.8" Called-Station-Id = "sip:8882@202.57.32.8" Sip-Translated-Request-URI = "sip:8882@202.57.32.56" Acct-Session-Id = "D5869094-562D-40EF-9A6F-10013821279A(a)202.57.32.54" Sip-To-Tag = "63ebc3f7" Sip-From-Tag = "2848041947" Sip-Cseq = "3141" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "1194d390d6772b16" Timestamp = 1126498424 Mon Sep 12 12:13:44 2005 Acct-Status-Type = Start <----- Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 4 User-Name = "8437310(a)202.57.32.8" Calling-Station-Id = "sip:8437310@202.57.32.8" Called-Station-Id = "sip:8882@202.57.32.8" Sip-Translated-Request-URI = "sip:8882@202.57.32.56:5060" Acct-Session-Id = "D5869094-562D-40EF-9A6F-10013821279A(a)202.57.32.54" Sip-To-Tag = "63ebc3f7" Sip-From-Tag = "2848041947" Sip-Cseq = "3141" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "1194d390d6772b16" Timestamp = 1126498424 Mon Sep 12 12:13:54 2005 Acct-Status-Type = Stop <------- Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 8 User-Name = "8882(a)202.57.32.8" Calling-Station-Id = "sip:8882@202.57.32.8" Called-Station-Id = "sip:8437310@202.57.32.8" Sip-Translated-Request-URI = "sip:8437310@202.57.32.54" Acct-Session-Id = "D5869094-562D-40EF-9A6F-10013821279A(a)202.57.32.54" Sip-To-Tag = "2848041947" Sip-From-Tag = "63ebc3f7" Sip-Cseq = "104" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "a1c74a53a5e1ed85" Timestamp = 1126498434 ______________________________________________________ Yahoo! for Good Watch the Hurricane Katrina Shelter From The Storm concert http://advision.webevents.yahoo.com/shelter

1 0

[Serusers] SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!!
by GT 12 Sep '05

12 Sep '05

Hello, I am trying SER with Mediaproxy 1.2.1 as per Onsip sample config. I am running SER & Mediaproxy on the same host rtp port range default (35000:65000) but even for non-nated devices I get no audio until I stop completely mediaproxy Debugging but thinking there might be something wrong with this config, any hint would be much appreciated, Thanks, Gtel

1 0

[Serusers] call me
by harry gaillac 11 Sep '05

11 Sep '05

hello, Please to call me at for tests. sip:84@nxs;yi.org or sip:85@nxs;yi.org or sip:86@nxs;yi.org or sip:87@nxs;yi.org Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Compilation Error on Mac OS (Darwin) - SEMS..
by Amos Nungu 11 Sep '05

11 Sep '05

I'm installing SEMS-0.9 on Mac OS (Darwin 8.2.0). upon running the make all command, I got an error as shown below ------------------------------------------------- # make all make[1]: Nothing to be done for `deps'. g++ -o sems AmApi.o AmAudio.o AmCmd.o AmConfig.o AmEventQueue.o AmIcmpWatcher.o AmMail.o AmPlugIn.o AmRequest.o AmRtpPacket.o AmRtpPacketTracer.o AmRtpScheduler.o AmRtpStream.o AmSdp.o AmServer.o AmSession.o AmSmtpClient.o AmThread.o AmUtils.o EmailTemplate.o SemsConfiguration.o SerClient.o SerDBQuery.o log.o sems.o -lm -ldl -lsocket -lnsl -lpthread /usr/bin/ld: can't locate file for: -lsocket collect2: ld returned 1 exit status make[1]: *** [sems] Error 1 make: [all] Error 2 (ignored) ------------------------------------------------------- Any idea on how to go about it will be great. Regards

1 0

[Serusers] I cannot compile "check_from()" method although I load module "uri"
by tulay sazak 11 Sep '05

11 Sep '05

Hello everbody, I loaded module "uri" and my ser.cfg is as below: ..... loadmodule "/usr/local/lib/ser/modules/uri.so" ..... if (method=="INVITE") { if (!radius_proxy_authorize("iptel.org")) { proxy_challenge("iptel.org", "1"); break; } if (!check_from()) { sl_send_reply("403", "Forbidden"); break; } } ........ When I run ser, it says: 0(10457) find_export: <check_from> not found 0(10457) find_export: <check_from> not found 0(10457) parse error (183,35-36): unknown command, missing loadmodule? Can you see what am I doing wrong ?? Thanks in advance Regards Tulay ______________________________________________________ Yahoo! for Good Watch the Hurricane Katrina Shelter From The Storm concert http://advision.webevents.yahoo.com/shelter

1 0

[Serusers] Media Relaying via MediaProxy
by Ozan Blotter 10 Sep '05

10 Sep '05

Hi List, Case : 1. Have a database , it include SIP no's, usernames, names & surnames... 2. Register myself as admin and load a voice message in wav format... 3. Need to deliver that voice message to everyone one by one via database... 4. If callee is busy then try it a few minutes later again... 5. Close calls... Does someone interested something like this by using RTP Relaying ? If yes, please tell me your ideas.. Thanks, Ozan

2 1

[Users] problems using cpl-c in combiantion with postgres DB
by Steen, Gerd 10 Sep '05

10 Sep '05

Hi, try to use cpl-c in combination with postgres. Running in several problems. The postgres database does not know a blob. I changed this to bytea. Next problem was the column "user" in the cpl table. I changed it to username and made changes in source files. After all this, I tried to upload a xml file with cple. The result where many error messages at insert of cpl table. Is there an idea? Regards Gerd

2 1

[Serusers] Serweb Click-to-dial Web-based application
by Frank Kostin 10 Sep '05

10 Sep '05

Hello All, Anyone implemented Click-to-dial Web-based application ? Seems that you need a phone with REFER support but what phone and how to configure ? I've been trying pingtel softphone without success. Any hint ? Thanks, Frank --------------------------------- Click here to donate to the Hurricane Katrina relief effort.

1 0

[Serusers] we are sending your missed calls
by fedora＠voice.dyndns.tv 10 Sep '05

10 Sep '05

Hello, we are sending your missed calls calling subscribertimereply status sip:8000@voice.dyndns.tv:5060 2005-09-09 05:57:52 400 Bad Request

1 0

RE: [Serusers] RFC3621
by harry gaillac 10 Sep '05

10 Sep '05

How can I avoid relay wiht this block Harry --- Zeus Ng <zeus.ng(a)isquare.com.au> a écrit : > Harry, > > You are detecting nat only within this block. > > if (!is_uri_host_local() || !is_from_local() ) { > nat detection ... > } > > This means any request that is not originate from > our domain or not for our > domain, detect nat. Now, when someone outside send > an INVITE to you, it will > pass this block, no nat detection, as it is for your > domain. Besides, you > are setting up a open relay. Anyone can use your > server to call someone > else. > > We usually send a 404 response within that block. > > I suggest you move the nat detection to the very > beginning, just after > checking max forward and message size. > > Zeus > > -----Original Message----- > From: harry gaillac [mailto:gaillacharry@yahoo.fr] > Sent: Friday, 9 September 2005 6:47 PM > To: Zeus Ng > Subject: RE: [Serusers] RFC3621 > > > Hello, > > Thanks for reply. > here are my ser.cfg and two ethereal files . > > You can see I use the latest mediaproxy (1.4.2) > > look at internal_sip_call file line 15-18 the > callee > send the public ip in SDP in reply. > > Look at incoming_call file line 9-10 the calle send > its private ip in SDP in reply. > > This is my problem !! > > Regards > Harry > > > Nothing is wrong with the reply. SER just forwards > > what the UA report its > > IP. > > > > I guest you are using a dual home server to run > SER > > and does not have > > routing logic to handle internal/external RTP > > communication. You need to > > bridge the RTP with either rtpproxy or mediaproxy. > I > > know the theory but > > haven't tried myself. The rtpproxy has some > example > > config for you to start. > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] sl_send_reply: I won't send a reply for ACK!! - Problem Identified
by Corey S. McFadden 10 Sep '05

10 Sep '05

Guys, I was able to figure out why I was getting occasional sl_send_reply errors with some ACK messages. I don't know if this is something that needs to be corrected at a deeper level, but I'll give you the rundown: 1. UA Makes a call to a bad number 2. Bad number is actually valid so it gets routed to the GW 3. GW responds 404 4. ACK loop occurs. When the ACK loop happens, SIP captures reveal the following (watch the Max-Forwards): #1 .. Max-Forwards: 9 Record-Route: <sip:ser;ftag=0014a934057d001145ed0fa2-6f6f2e8e;lr=on> Via: SIP/2.0/UDP ser;branch=0 Via: SIP/2.0/UDP 192.168.249.83:5060;received=otherip;branch=z9hG4bK140d833e .. #2 .. Record-Route: <sip:ser;ftag=0014a934057d001145ed0fa2-6f6f2e8e;lr=on> Max-Forwards: 8 Record-Route: <sip:ser;ftag=0014a934057d001145ed0fa2-6f6f2e8e;lr=on> Via: SIP/2.0/UDP ser;branch=0 Via: SIP/2.0/UDP ser;branch=0 Via: SIP/2.0/UDP 192.168.249.83:5060;received=otherip;branch=z9hG4bK140d833e .. Etc. etc. until Max Forwards counts down to 0, at which point: if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; And naturally: sl_send_reply: I won't send a reply for ACK!! So, I've obviously been able to avoid this by changing to: if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); }; break; }; Now, the greater question: is the ACK loop being generated by a defective message. If so, is it anything to worry about? And in either case is putting the method check in there a valid solution? Can anyone advise? Thanks, -Corey ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

3 2

[Serusers] One failure_route problem remaining
by Corey S. McFadden 10 Sep '05

10 Sep '05

Guys, First, thanks again to those who have offered advice. I have re-started building from the ONsip.org rtpproxy example config and have everything working properly except one thing. In the interest of being thorough, I kept a log of what I changed and why: - Tested REGISTER, works OK. - Calls between UAs initially failed when proxy_authorize in route[3] was called. Disabled proxy_authorize checks. Works OK. (Need to revisit this one for security. Can anyone point out the problem?) - Implemented failure_route for voicemail on timeout. Timeout calls from UAs work fine to VM via Asterisk. - Implemented 7 digit dial perl script and added second lookup. No problems. Added NPANXX route lookup script in route[3]. No problems. - Implemented "Your call cannot be completed as dialed" via Asterisk server by replacing sl_send_reply 404 in route[3] with route(7). No problems. - Initially was unable to dial voicemail access number from UA. It would connect to Asterisk server but not have audio. Call would drop. Fixed by adding this to route[1]: # NAT Fix if (method=="INVITE" && nat_uac_test("19")) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; - Voicemail access is now OK. - UA timeout calls to voicemail now fail. Debug on Asterisk looks like the contact is being fixed twice. Added flag(9) to make sure NAT fix only happens once. UA timeout calls now go to voicemail properly. - I notice a lot of "Warning: sl_send_reply: I won't send a reply for ACK!!" but don't know if this is significant or not. From what I've read it sounds like ACKs are getting an sl_reply rather than being t_relayed but I didn't really modify anything related ... ? Now, the biggest problem seems to be calls coming from the inbound SIP gateway aren't being handled properly on timeout. (Calls from the inbound SIP gateway work fine to UAs.) This very well may be the fault of the other equipment, but it's nothing we can debug directly so I want to make sure there's nothing else wrong. Keep in mind, the inbound SIP gateway CAN communicate with the Asterisk server by dialing a DID that is directed to voicemail. I thought it might be some kind of routing problem between those boxes, but this would seem to disprove that. I'm going to include the abbreviated ngrep dump from the SER machine and then the ser.cfg file. Thanks again for the help thus far and in advance for any assistance on this. -Corey Here is an IP legend: xxx.xxx.xxx.7 - Inbound SIP Gateway xxx.xxx.xxx.36 - SER Server xxx.xxx.xxx.80 - Asterisk Server 216.xxx.ua.207 - NATed UA public IP ** NGREP DUMP FROM SER SERVER # U xxx.xxx.xxx.7:5060 -> xxx.xxx.xxx.36:5060 INVITE sip:5414362504@xxx.xxx.xxx.36:5060 SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41 # U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565 INVITE sip:5414362504@216.xxx.ua.207:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=Xvn755Ml3U1j41B-IPTrunk-67-17-23at # U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0,SIP/2.0/UDP xxx.xxx.xxx.7:5060;rport=5060 # U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0,SIP/2.0/UDP xxx.xxx.xxx.7:5060;rport=506 # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;rport=5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..From: sip:61087 # # *** Call is diverted to Asterisk server here. # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.80:5060 INVITE sip:5414362504@xxx.xxx.xxx.80:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=Xvn755Ml3U1j41B-IPTrunk-67-17-23a # U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565 CANCEL sip:5414362504@216.xxx.ua.207:5060 SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0..From: sip: # U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;bran # U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0..From: sip:6108712381@xxx.xxx.xxx.7:5060;tag= # U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060 SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0,SIP/2.0/UDP xxx.xxx.xxx.7:5060 # U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565 ACK sip:5414362504@216.xxx.ua.207:5060 SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0..From: sip:610 # U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:5060 .... # U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565 .... # U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route: <sip:209.216.160. # U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route: <sip:209.216.160. # # Several OKs seem to be going through here... U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route: <sip:209.216.160. # U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route: <sip:209.216.160. # # Call is dropped by now. ** Asterisk console sip debug <-- SIP read from xxx.xxx.xxx.36:5060: INVITE sip:5414362504@xxx.xxx.xxx.80:5060 SIP/2.0 Record-Route: <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on> Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1 Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;rport=5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0 Max-Forwards: 5 From: sip:6108712381@xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7 To: sip:5414362504@xxx.xxx.xxx.36:5060 Call-ID: AJn351hl3U1H58O(a)xxx.xxx.xxx.7 CSeq: 101 INVITE Remote-Party-Id: <sip:6108712381@xxx.xxx.xxx.7:5060>;party=calling;privacy=off;id-type=subscriber Expires: 180 Allow: INVITE,CANCEL,BYE,ACK Contact: sip:xxx.xxx.xxx.7:5060 User-Agent: Tekelec-7000/r4.0 Content-Type: application/sdp Content-Length: 149 P-hint: OFFLINE VOICEMAIL v=0 o=- 1 1 IN IP4 xxx.xxx.xxx.7 s= c=IN IP4 xxx.xxx.xxx.36 t=0 0 m=audio 35842 RTP/AVP 0 a=ptime:20 a=rtpmap:0 PCMU/8000 a=nortpproxy:yes --- (17 headers 9 lines)--- Using INVITE request as basis request - AJn351hl3U1H58O(a)xxx.xxx.xxx.7 Sending to xxx.xxx.xxx.36 : 5060 (non-NAT) Found no matching peer or user for 'xxx.xxx.xxx.36:5060' Found RTP audio format 0 Peer audio RTP is at port xxx.xxx.xxx.36:35842 Found description format PCMU Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 5414362504 in sip list_route: hop: <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on> list_route: hop: <sip:xxx.xxx.xxx.7:5060> Transmitting (no NAT) to xxx.xxx.xxx.36:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1 Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0 From: sip:6108712381@xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7 To: sip:5414362504@xxx.xxx.xxx.36:5060 all-ID: AJn351hl3U1H58O(a)xxx.xxx.xxx.7 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:5414362504@xxx.xxx.xxx.80> Content-Length: 0 --- -- Executing Wait("SIP/5060-b7d05f68", "1") in new stack -- Executing VoiceMail("SIP/5060-b7d05f68", "5414362504@sip") in new stack We're at xxx.xxx.xxx.80 port 14018 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x400 (ilbc) Reliably Transmitting (no NAT) to xxx.xxx.xxx.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1 Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0 Record-Route: <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on> From: sip:6108712381@xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7 To: sip:5414362504@xxx.xxx.xxx.36:5060;tag=as4c6bfaee Call-ID: AJn351hl3U1H58O(a)xxx.xxx.xxx.7 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:5414362504@xxx.xxx.xxx.80> Content-Type: application/sdp Content-Length: 237 v=0 o=root 29275 29275 IN IP4 xxx.xxx.xxx.80 s=session c=IN IP4 xxx.xxx.xxx.80 t=0 0 m=audio 14018 RTP/AVP 0 3 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - --- -- Playing 'vm-intro' (language 'en') Retransmitting #1 (no NAT) to xxx.xxx.xxx.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1 Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0 Record-Route: <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on> From: sip:6108712381@xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7 To: sip:5414362504@xxx.xxx.xxx.36:5060;tag=as4c6bfaee Call-ID: AJn351hl3U1H58O(a)xxx.xxx.xxx.7 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:5414362504@xxx.xxx.xxx.80> Content-Type: application/sdp Content-Length: 237 v=0 o=root 29275 29275 IN IP4 xxx.xxx.xxx.80 s=session c=IN IP4 xxx.xxx.xxx.80 t=0 0 m=audio 14018 RTP/AVP 0 3 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - --- ** ser.cfg # whatever.net - SER Configuration File # debug=3 fork=yes log_stderror=yes listen=xxx.xxx.xxx.36 port=5060 children=4 alias=whatever.net alias=hostname.whatever.net dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:serpass@localhost/ser" # Modules loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/options.so" loadmodule "/usr/local/lib/ser/modules/exec.so" modparam("auth_db|uri_db|usrloc|permissions", "db_url", "mysql://ser:serpass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 20) modparam("tm", "fr_timer", 20) # # Route Section # route { # Basic Checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; ## # Record Route Section ## if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } ## # Loose Route Section ## if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; ## # Message Processing Section ## if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="NOTIFY") { sl_send_reply("200", "Shut up."); break; } else if (method=="OPTIONS" && (! uri=~"sip:.*[@]+.*")) { options_reply(); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location") && method!="ACK") { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # NAT Fix if (method=="INVITE" && nat_uac_test("19") && !isflagset(9)) { xlog("L_ERR", "Fixing NAT on INVITE for %ct.\n"); force_rport(); fix_nated_contact(); force_rtp_proxy(); setflag(9); }; t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("whatever.net","subscriber")) { www_challenge("whatever.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:911@") { # Pass 911 calls first route(5); break; }; if (uri=~"^sip:[0-9]{7}@") { # EXPAND 7-DIGIT CALL exec_dset("/root/7digit.pl"); }; # Try location lookup again if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # DB Lookup of NPANXX route exec_dset("/root/npa_nxx.pl"); route(1); break; }; if (uri=~"^sip:[0-9]{11}@") { # DB Lookup of NPANXX route strip(1); exec_dset("/root/npa_nxx.pl"); route(1); break; }; if (uri=~"^sip:555@") { # Voicemail Access route(6); break; }; if (uri=~"^sip:556@") { # Voicemail Access route(6); break; }; # Call error message route(7); # sl_send_reply("404", "User Not Found"); break; }; }; # if (!proxy_authorize("whatever.net","subscriber")) { # proxy_challenge("whatever.net","0"); # break; # } else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; # }; # consume_credentials(); if (isflagset(6) && !isflagset(9)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); setflag(9); }; t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[5] { ### # Local PSTN Gateway ### rewritehost("xxx.xxx.xxx.7"); route(1); } route[6] { ### # Calls to VM ### rewritehost("xxx.xxx.xxx.80"); route(1); } route[7] { ### # Call Failure ### rewriteuri("sip:7110000001@xxx.xxx.xxx.80"); route(1); } failure_route[1] { ### # Voicemail ### xlog("L_ERR", " Failure Route 1 hit.\n"); if (t_check_status("487")) { xlog("L_ERR", " Failure Route 1 - 487 detected, breaking.\n"); break; }; rewritehost("xxx.xxx.xxx.80"); append_hf("P-hint: OFFLINE VOICEMAIL\r\n"); append_branch(); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

2 6

[Serusers] Can I use exec commands for authentication?
by Ryan Pagquil 10 Sep '05

10 Sep '05

Hi, I'm thinking of implementing authentication using exec commands. For example, I have a script that checks if a user exists in a database. For Register methods i will run this script, if the script exit with 1 the use will be rejected and if the script exit 0 the user is authenticated. Is this possible? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 1

[Serusers] Media Path
by Aster 09 Sep '05

09 Sep '05

Hello all, I would like to understand the mediapath for various setup. Please help me to understand how media (voice/video) travels between UAC1 and UAC2 for different setup. These by assuming mediaproxy/rtpproxy running on SER. And will there be any difference in the media path if record_route is used? 1) UAC1 (Public IP)----------SER(Public IP)-----------UAC2 (Public IP) 2) UAC1 (Private IP)----------SER(Public IP)-----------UAC2 (Public IP) 3) UAC1 (Private IP)----------SER(Public IP)-----------UAC2 (Private IP) 4) UAC1 (Private IP)----------SER(Private IP)-----------UAC2 (Public IP) 5) UAC1 (Private IP)----------SER(Private IP)-----------UAC2 (Private IP) 6) UAC1 (Public IP)----------SER(Private IP)-----------UAC2 (Public IP) TIA

1 0

[Serusers] Asterisk "clustering" with a SER proxy?
by Roy Sigurd Karlsbakk 09 Sep '05

09 Sep '05

hi i've heard it should be possible, but i can't find out how... I want to configure a bunch of asterisk boxes to do SIP/PSTN connectivity, and I need SER or something to do some balancing in front of them. The requirements are listed below. * SER MUST accept and load balance incoming calls over n asterisk boxes (anything between 2 and 20 servers depending on installation) * If SER forwards a call to a server being busy or down, SER SHOULD retry on another server * SER SHOULD balance the number of calls to each server based on the codec used so single servers will not be overloaded by transcoding costs. * If possible, SER SHOULD be able to fail over to another SER box if SER fails. Does anyone know if this is possible? I'd gladly pay someone to help me out here... roy

7 7

[Serusers] Multiple SER Problems
by Corey S. McFadden 09 Sep '05

09 Sep '05

Hi everyone, We're working through various SER problems and my thinking is that we've got something generally wrong with our SER config. I posted previously about one of these problems but wanted to clarify the situation and see if anyone could offer some advice. Network: +-------------+ +--| Asterisk VM | | +-------------+ +---------------+ +---+ | +----------------+ |Inbound PSTN GW|----|SER|--+--| Outbound LD GW | +---------------+ +---+ | +----------------+ | | +------------------+ +-------+ | +--| Outbound PSTN GW | | UAs |-----+ +------------------+ +-------+ | | +-----------+ +--+ | | NATed UAs |--|FW|--+ +-----------+ +--+ Problems: - All UAs can make outbound calls without a problem. - UAs outside NAT can call each other. - NATed UAs can call each other. Depending on the UA it will either get one way audio or no audio. (Cisco 7960 seems to have more problems than the Sipura 841) - PSTN GW calls to NATed UAs ring but can't answer the call. SER logs: Warning: sl_send_reply: I won't send a reply for ACK!! - When PSTN calls a UA and it doesn't answer, failure route fails to connect the call to Asterisk. The call is set up (Asterisk logs attempts to send OKs back and will launch the voicemail app) but the PSTN gateway never gets the OK. Call is lost. - When NATed UAs call each other without answering fr_invite_timer hits and failure route fails to send the call to Asterisk. I've read a LOT of back postings on the list and have read a bunch of threads that describe the one-way audio problem but never could find any acutal answers or resolution. The threads just seem to end. (i.e. http://lists.iptel.org/pipermail/serusers/2004-October/012305.html ) I'm going to attach the ser.cfg but will hold off on the ngrep and related documentation as I think someone might be able to tell me what we're doing wrong by just looking at this. Anyhow, if anyone can offer some insight it would be greatly appreciated. (If anyone who is reading this is interested in consulting on this problem, contact me off-list and we can make payment arrangements.) Thanks in advance, -Corey SER Config: ############################################################ # # whatever.net - SER Configuration File # ser.conf # Revised 7/31/2005 # ############################################################ debug=3 fork=yes log_stderror=yes listen=xxx.ser.ip.addr port=5060 children=4 alias=whatever.net alias=hostname.whatever.net dns=no rev_dns=no # Added 9/3 reply_to_via=no check_via=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:passwd@localhost/ser" ### # Modules ### loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" ### # Module Paramaters ### modparam("auth_db|uri_db|usrloc|permissions", "db_url", "mysql://ser:passwd@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 5) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # VM Timeout #modparam("tm", "fr_inv_timer", 20) modparam("tm", "fr_inv_timer", 10) modparam("tm", "fr_timer", 10) modparam("usrloc", "db_mode", 2) modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("exec", "setvars", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_url", "mysql://ser:passwd@localhost/ser") modparam("acc", "db_flag", 1) ### # Routing Section ### route { # Basic Checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; if (method=="INVITE" && nat_uac_test("19")) { xlog("L_ERR", "INVITE detected with nat_uac_test 19. Adding record route preset.\n"); record_route_preset("xxx.ser.ip.addr:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # Loose Routing if (loose_route()) { route(1); break; }; # Call Processing if (uri!=myself) { log (1, "*** Message URI is not myself. Probably should not happen.\n"); route(1); break; }; if (uri==myself) { if (method=="ACK") { xlog("L_ERR", "ACK received from %is for %tu.\n"); route(1); break; } else if (method=="CANCEL") { xlog("L_ERR", "CANCEL received from %is for %tu.\n"); log(1, " Cancel\n"); route(3); break; } else if (method=="INVITE") { xlog("L_ERR", "INVITE received from %is for %tu.\n"); route(3); break; } else if (method=="REGISTER") { xlog("L_ERR", "REGISTER received from %is.\n"); route(2); break; } else if (method=="NOTIFY") { sl_send_reply("200", "Shut up."); break; } else if (method=="OPTIONS") { sl_send_reply("200", "Shut up."); break; }; }; route(1); } route[1] { ### # Default Call Handling ### # NAT Fix if (method=="INVITE" && nat_uac_test("19") && !isflagset(9)) { xlog("L_ERR", " - INVITE NAT detected (route 1).\n"); xlog("L_ERR", " -- Contact Header: %ct\n"); fix_nated_sdp("1"); force_rport(); fix_nated_contact(); force_rtp_proxy(); setflag(9); xlog("L_ERR", " ++ Contact Header: %ct\n"); }; # On call timeout to voicemail # t_on_failure("1"); # t_on_reply("1"); # Log setflag(1); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { ### # REGISTER method ### log(1," Route 2 hit.\n"); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("whatever.net","subscriber")) { www_challenge("whatever.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { ### # INVITE & CANCEL # URI matching is here. ### xlog("L_ERR", " Route 3 starting.\n"); t_on_failure("1"); t_on_reply("1"); if (nat_uac_test("19")) { setflag(6); }; if (method=="INVITE" && !allow_trusted()) { # Proxy auth code goes here consume_credentials(); log(1,"INVITE and not trusted."); }; lookup("aliases"); if (!lookup("location")) { if (uri=~"^sip:911@") { # Pass 911 calls first route(5); break; }; if (uri=~"^sip:[0-9]{7}@") { # EXPAND 7-DIGIT CALL exec_dset("/root/7digit.pl"); }; # Try location lookup again if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # DB Lookup of NPANXX route exec_dset("/root/npa_nxx.pl"); route(1); break; }; if (uri=~"^sip:1[0-9]{10}@") { # DB Lookup of NPANXX route exec_dset("/root/npa_nxx.pl"); route(1); break; }; if (uri=~"^sip:555@") { # Voicemail Access route(6); break; }; if (uri=~"^sip:556@") { # Voicemail Access route(6); break; }; # Call error message route(7); break; }; }; # NAT Fix if (nat_uac_test("19") && !isflagset(9)) { fix_nated_sdp("1"); force_rport(); fix_nated_contact(); force_rtp_proxy(); setflag(9); }; if (!t_relay()) { if(nat_uac_test("19") && isflagset(9)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { ### # Primary LD Gateway ### rewritehost("xxx.sip.gw.ip"); route(1); } route[5] { ### # Local PSTN Gateway ### rewritehost("xxx.pstn.gw.ip"); route(1); } route[6] { ### # Calls to VM ### rewritehost("xxx.ast.srv.ip"); route(1); } route[7] { ### # Call Failure ### rewriteuri("sip:7110000001@xxx.ast.srv.ip"); route(1); } failure_route[1] { ### # Voicemail ### xlog("L_ERR", " * Failure Route 1 "); rewritehost("xxx.ast.srv.ip"); append_hf("P-hint: OFFLINE VOICEMAIL\r\n"); append_branch(); route(1); } onreply_route[1] { if (nat_uac_test("19") && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1") && !isflagset(9)) { fix_nated_contact(); }; } END ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

3 3

[Serusers] Caller ID
by Sebastian Kühner 09 Sep '05

09 Sep '05

Hi! Is there any way to change the caller id in the PSTN route? If I make a call to a PSTN phone, the display of this phone shows the internal username. It would be nice to show a Virtual Number or something like that. Thanks! Sebastian

4 5

[Serusers] Question about AUTH MODULE
by Ricardo Martinez 09 Sep '05

09 Sep '05

Hello list. I was wondering if the module auth_radius can receive extra parameters from the Radius Server and proccess them as AVP values (something like the avp_radius can do) . I'm using : version: ser 0.9.4-rc1 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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 04:26:34 Jul 21 2005 with gcc 3.2 I was browsing the CVS directory and i noticed that this revision : Revision 1.18 / (view) - annotate - [select for diffs] , Thu Jun 30 20:29:31 2005 UTC (2 months ago) by andrei Branch: MAIN CVS Tags: last_merge_to_janakj, HEAD Changes since 1.17: +6 -2 lines Diff to previous 1.17 seems to do what i want. Am i right? Can the version that i'm actually using do the same? I Hope that somone can help me. Thanks! Regards, Ricardo Martinez.-

1 0

[Serusers] SER and OpenSER
by Waldo Rubinstein 09 Sep '05

09 Sep '05

Can anyone tell me what the difference between SER and OpenSER is? Thanks, Waldo

5 5

[Serusers] Issues when using LCR module
by Anthony Dean 09 Sep '05

09 Sep '05

I am getting a weird issue when I use the LCR modules and I am not sure if its a bug in the module or my config. It only happens when the first leg has failed the invite and it jumps to the failure route and tries the next gateway on the list. The bug is in the "c=IN IPV4" line in the invite to the second gateway where there should only be one ip address. So it sends the following: c=IN IPV4 192.168.8.35192.168.8.35. when it should be: c=IN IPV4 192.168.8.35. This makes the second gateway give back a 488 not acceptable back regardless if it can process the call or not as the header is malformed. If the first gateway of choice doesn't give back an error the call will connect just fine as the first invite is correctly formatted (which makes me think it is something to do with how I process the failure route that is at fault or how the lcr does it - just can't figure out which). Below is the sip message flow that I get (I made the first gateway give back a 404 not found to test the lcr failover). Call flow is supposed to be: PH-A = 2958770000 (Caller, 192.168.10.30 [Nat Inside], 192.168.7.250 [Nat Outside]) PH-B = 2959770000 (Called, PSTN) GW-A = 192.168.8.6 GW-B = 192.168.35.18 SER = 192.168.8.35 PH-A -> SER (INVITE - Good) PH-A <- SER (100 Trying - Good) SER -> GW-A (INVITE - Good) SER <- GW-A (100 Trying - Good) SER <- GW-A (404 Not Found - Good) SER -> GW-A (ACK - Good) SER -> GW-B (INVITE - Malformed) SER <- GW-B (488 Not Acceptable - Due To Malformed Header) ... Then the call fails due to no more gateways as expected. Any ideas what is causing this (my config is mainly based on the onsip pstn example with lcr extras)? U 192.168.7.250:61903 -> 192.168.8.35:5060 INVITE sip:2959770000@192.168.8.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Max-Forwards: 70. Contact: 2958770000 <sip:2958770000@192.168.10.30:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 286. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 712750 712750 IN IP4 192.168.10.30. s=-. c=IN IP4 192.168.10.30. t=0 0. m=audio 16430 RTP/AVP 18 0 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.8.35:5060 -> 192.168.7.250:61903 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765;rport=61903;received=192.168.7.25 0. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Server: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.8.6:5060 INVITE sip:12959770000@192.168.8.6 SIP/2.0. Record-Route: <sip:2959770000@192.168.8.35:5060;nat=yes;ftag=d0ff6100e33025ado0;lr=on>. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0. Via: SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 2958770000 <sip:2958770000@192.168.7.250:61903>. Expires: 240. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 284. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 712750 712750 IN IP4 192.168.10.30. s=-. c=IN IP4 192.168.8.35. t=0 0. m=audio 35040 RTP/AVP 18 0 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.8.6:50183 -> 192.168.8.35:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0,SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=22C61F2C-2339. Date: Thu, 08 Sep 2005 21:06:38 GMT. Call-ID: c19d05bc-82926821(a)192.168.10.30. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Length: 0. . U 192.168.8.6:50183 -> 192.168.8.35:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0,SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=22C61F2C-2339. Date: Thu, 08 Sep 2005 21:06:38 GMT. Call-ID: c19d05bc-82926821(a)192.168.10.30. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Reason: Q.850;cause=3. Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.8.6:5060 ACK sip:12959770000@192.168.8.6 SIP/2.0. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. Call-ID: c19d05bc-82926821(a)192.168.10.30. To: <sip:2959770000@192.168.8.35>;tag=22C61F2C-2339. CSeq: 101 ACK. User-Agent: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.35.18:5060 INVITE sip:12959770000@192.168.35.18 SIP/2.0. Record-Route: <sip:2959770000@192.168.8.35:5060;nat=yes;ftag=d0ff6100e33025ado0;lr=on>. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.1. Via: SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 2958770000 <sip:2958770000@192.168.7.250:61903>. Expires: 240. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 300. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 712750 712750 IN IP4 192.168.10.30. s=-. c=IN IP4 192.168.8.35192.168.8.35. t=0 0. m=audio 3504035040 RTP/AVP 18 0 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.35.18:5060 -> 192.168.8.35:5060 SIP/2.0 488 Not acceptable here. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.1. Via: SIP/2.0/UDP 192.168.10.30:5060;received=192.168.7.250;branch=z9hG4bK-701a9765. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=as113c57f8. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. User-Agent: Video PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:12959770000@192.168.35.18>. Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.35.18:5060 ACK sip:12959770000@192.168.35.18 SIP/2.0. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.1. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. Call-ID: c19d05bc-82926821(a)192.168.10.30. To: <sip:2959770000@192.168.8.35>;tag=as113c57f8. CSeq: 101 ACK. User-Agent: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.7.250:61903 SIP/2.0 503 Service Temporarilly Unavailable, No More Available Gateways. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765;rport=61903;received=192.168.7.25 0. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=d1bd3b96fe18e5f9baf4949f8e4468f7-0b42. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Server: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.7.250:61903 -> 192.168.8.35:5060 ACK sip:2959770000@192.168.8.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=d1bd3b96fe18e5f9baf4949f8e4468f7-0b42. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 ACK. Max-Forwards: 70. Contact: 2958770000 <sip:2958770000@192.168.10.30:5060>. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 0. . # ----------------------------------------------------------------- # Standard Settings # ----------------------------------------------------------------- debug = 3 children = 1 fork = no log_stderror = yes check_via = no dns = no rev_dns = no sip_warning = no # ----------------------------------------------------------------- # Server Settings # ----------------------------------------------------------------- port = 5060 listen = 192.168.8.35 alias = 192.168.8.35 # ----------------------------------------------------------------- # Fifo Settings # ----------------------------------------------------------------- fifo = "/tmp/ser_fifo" fifo_mode = 0666 fifo_db_url = "mysql://root:password@192.168.8.35/ser" # ----------------------------------------------------------------- # Module Settings # ----------------------------------------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/auth_radius.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/uri_radius.so" loadmodule "/usr/lib/ser/modules/uac.so" loadmodule "/usr/lib/ser/modules/lcr.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/group_radius.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/avp_radius.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/options.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/speeddial.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------------------------------------------------------- # Module Parameters # ----------------------------------------------------------------- modparam("auth", "secret", "password") modparam("auth", "nonce_expire", 60) modparam("auth", "rpid_prefix", "") modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "realm_prefix", "sip.") modparam("auth", "rpid_avp", "i:13") modparam("auth_db", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("uri_db", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("uri_radius", "service_type", 15) modparam("domain", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("domain", "db_mode", 0) modparam("domain", "domain_table", "domain") modparam("group", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") modparam("group", "use_domain", 0) modparam("group_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("group_radius", "use_domain", 0) modparam("registrar", "nat_flag", 6) modparam("registrar", "append_branches", 0) modparam("registrar", "case_sensitive", 0) modparam("registrar", "default_expires", 300) modparam("registrar", "default_q", 100) modparam("registrar", "max_contacts", 0) modparam("registrar", "max_expires", 300) modparam("registrar", "min_expires", 60) modparam("registrar", "use_domain", 0) modparam("registrar", "received_avp", 42) modparam("registrar", "received_param", "received") modparam("registrar", "retry_after", 0) modparam("permissions", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions/allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions/deny") modparam("permissions", "check_all_branches", 1) modparam("permissions", "allow_suffix", ".allow") modparam("permissions", "deny_suffix", ".deny") modparam("usrloc", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "use_domain", 0) modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "method_column", "method") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("avpops", "avp_url", "mysql://root:password@192.168.8.35/ser") modparam("avpops", "avp_aliases", "voicemail=i:500") modparam("avpops", "avp_table", "preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("lcr", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 30) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "rpid") modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 30) modparam("tm", "wt_timer", 10) modparam("tm", "delete_timer", 2) modparam("tm", "retr_timer1p1", 1) modparam("tm", "retr_timer1p2", 2) modparam("tm", "retr_timer1p3", 4) modparam("tm", "retr_timer2", 4) modparam("tm", "noisy_ctimer", 0) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "fr_timer_avp", "callee_fr_timer") modparam("tm", "fr_inv_timer_avp", "callee_fr_inv_timer") modparam("tm", "tw_append", "") modparam("acc", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("acc", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "service_type", 15) modparam("acc", "log_flag", 21) modparam("acc", "log_missed_flag", 22) modparam("acc", "db_flag", 23) modparam("acc", "db_missed_flag", 24) modparam("acc", "radius_flag", 25) modparam("acc", "radius_missed_flag", 26) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "early_media", 0) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", 42) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "force_socket", "") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/ser/mediaproxy/sip") modparam("mediaproxy", "rtp_asymmetrics", "/etc/ser/mediaproxy/rtp") modparam("mediaproxy", "natping_interval", 30) modparam("exec", "setvars", 1) modparam("speeddial", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("speeddial", "use_domain", 0) modparam("speeddial", "user_column", "username") modparam("speeddial", "domain_column", "domain") modparam("speeddial", "sd_user_column", "sd_username") modparam("speeddial", "sd_domain_column", "sd_domain") modparam("speeddial", "new_uri_column", "new_uri") modparam("speeddial", "domain_prefix", "sip.") modparam("maxfwd", "max_limit", 16) modparam("options", "accept", "*/*") modparam("options", "accept_encoding", "") modparam("options", "accept_language", "en") modparam("xlog", "buf_size", 8192) modparam("uac", "from_store_param", "vsf") modparam("uac", "credential", "username:domain:password") modparam("msilo", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("msilo", "db_table", "silo") modparam("msilo", "registrar", "sip:registrar@tristarcorp.net") modparam("msilo", "expire_time", 259200) modparam("msilo", "check_time", 30) modparam("msilo", "use_contact", 1) modparam("avp_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("avp_radius", "caller_service_type", 15) modparam("avp_radius", "callee_service_type", 15) # ----------------------------------------------------------------- # Main Route # ----------------------------------------------------------------- route { # ----------------------------------------------------------------- # Sanity Check # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method != "ACK") { sl_send_reply("483", "Too Many Hops"); }; break; }; if (msg:len > max_len) { if (method != "ACK") { sl_send_reply("513", "Message Overflow"); }; break; }; # ----------------------------------------------------------------- # Record Route # ----------------------------------------------------------------- if (method == "INVITE" && client_nat_test("3")) { record_route_preset("192.168.8.35:5060;nat=yes"); } else if (method != "REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down # ----------------------------------------------------------------- if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Routing # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method == "INVITE" || method == "ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Processing # ----------------------------------------------------------------- if (uri != myself) { route(1); break; }; if (uri == myself) { if (method == "REGISTER") { route(2); break; } else if (method == "REFER" || method == "INVITE" || method == "CANCEL") { route(3); break; } else if (method == "OPTIONS") { options_reply(); break; }; lookup("aliases"); if (uri != myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); break; } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { # ----------------------------------------------------------------- # Reply Routing # ----------------------------------------------------------------- t_on_reply("1"); # ----------------------------------------------------------------- # Call Log & Accounting (log=21, sql=23, radius=25) # ----------------------------------------------------------------- setflag(23); # ----------------------------------------------------------------- # Relay Call # ----------------------------------------------------------------- if (!t_relay()) { if (method == "INVITE" || method == "ACK") { end_media_session(); }; if (method != "REFER" && method != "INVITE" && method != "CANCEL" && method != "ACK") { sl_reply_error(); }; }; } # ----------------------------------------------------------------- # Register Message Handler # ----------------------------------------------------------------- route[2] { # ----------------------------------------------------------------- # Nat Test # ----------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; # ----------------------------------------------------------------- # Authenticate User # ----------------------------------------------------------------- if (!allow_trusted() && !from_gw()) { if (!www_authorize("", "subscriber")) { www_challenge("", "1"); break; }; consume_credentials(); }; # ----------------------------------------------------------------- # Save Registration # ----------------------------------------------------------------- if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # Invite/Cancel Message Handler # ----------------------------------------------------------------- route[3] { # ----------------------------------------------------------------- # Nat Fixup # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; # ----------------------------------------------------------------- # Authorize Users / Setup Voicemail For Gateways # ----------------------------------------------------------------- if (!allow_trusted() || !from_gw()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "1"); break; }; consume_credentials(); }; # ----------------------------------------------------------------- # Check For Alias # ----------------------------------------------------------------- lookup("aliases"); if (uri != myself) { route(4); break; }; # ----------------------------------------------------------------- # Cancel Call # ----------------------------------------------------------------- if (method == "CANCEL") { route(1); break; }; # ----------------------------------------------------------------- # Check For Local User # ----------------------------------------------------------------- lookup("location"); if (uri != myself) { route(4); break; }; # ----------------------------------------------------------------- # Route Call # ----------------------------------------------------------------- if (uri =~ "^sip:1[0-9]{10}@" || uri =~ "^sip:011") { route(5); break; } else { sl_send_reply("404", "User Not Found"); break; }; } # ----------------------------------------------------------------- # Media Proxy Enabler # ----------------------------------------------------------------- route[4] { # ----------------------------------------------------------------- # Enable Media Proxy # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); break; } # ----------------------------------------------------------------- # Gateway LCR Route Handler # ----------------------------------------------------------------- route[5] { # ----------------------------------------------------------------- # Load Pstn LCR Gateways # ----------------------------------------------------------------- if (!load_gws()) { sl_send_reply("500", "Internal Server Error, No PSTN Gateways"); break; }; # ----------------------------------------------------------------- # Find Next Available Gateway # ----------------------------------------------------------------- if (!next_gw()) { sl_send_reply("503", "Service Temporarilly Unavailable"); break; } # ----------------------------------------------------------------- # Route Call # ----------------------------------------------------------------- t_on_failure("1"); route(4); break; } # ----------------------------------------------------------------- # LCR Route Failure Handler # ----------------------------------------------------------------- failure_route[1] { # ----------------------------------------------------------------- # Failed Call Log & Accounting (log=22, sql=24, radius=26) # ----------------------------------------------------------------- setflag(24); # ----------------------------------------------------------------- # Find Next Available Gateway # ----------------------------------------------------------------- if (method == "INVITE" && t_check_status("404|408|500|503")) { if (!next_gw()) { t_reply("503", "Service Temporarilly Unavailable, No More Available Gateways"); break; }; t_on_failure("1"); route(4); break; }; } # ----------------------------------------------------------------- # Reply Route # ----------------------------------------------------------------- onreply_route[1] { # ----------------------------------------------------------------- # Enable Media Proxy # ----------------------------------------------------------------- if ((isflagset(6) || isflagset(7)) && (status =~ "(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; # ----------------------------------------------------------------- # Nat Fixup # ----------------------------------------------------------------- if (client_nat_test("1")) { fix_nated_contact(); }; } Thank You Anthony Dean This correspondence is considered confidential and any reproduction for the purpose of public disclosure is forbidden without written permission by the author signed above. If you are not the intended recipient, please immediately notify the sender and delete any copies. If you are responding to this e-mail, please append or attach all previous correspondence as a point of reference.

2 2

[Serusers] 503 service unavailable
by zhaomin 09 Sep '05

09 Sep '05

I use asterisk as a pstn and voicemail,but I get this error from asterisk 's CLI blow: Verbosity is at least 3 -- Executing AbsoluteTimeout("SIP/61.183.23.216-08a3b558", "15") in new stack -- Set Absolute Timeout to 15 -- Executing Congestion("SIP/61.183.23.216-08a3b558", "") in new stack == Spawn extension (from-sip-external, 2000, 2) exited non-zero on 'SIP/61.183.23.216-08a3b558' -- Executing AbsoluteTimeout("SIP/61.183.23.216-08a3b558", "15") in new stack -- Set Absolute Timeout to 15 -- Executing Congestion("SIP/61.183.23.216-08a3b558", "") in new stack == Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/61.183.23.216-08a3b558' What's wrong in it? ser or asterisk? I am not sure .anyones would help me? Thanks Zhaommin

1 0

[Serusers] one way voice on a public IP address
by Iqbal 09 Sep '05

09 Sep '05

Hi Has anyone had a problem like this, everything is on a public IP, even the UA, I get voice from UA --> PSTN, not the other way, the SDP all looks fine, c= has the IP address of the UA in it, I was thinking it might be a port problem, but the logs are not showing anything there. Iqbal

2 1

[Serusers] Problem with ACK's (branch=0)
by Rodrigo P. Telles 09 Sep '05

09 Sep '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've been experiencing some troubles with ACK's with branch=0. I found a thread about it but I didn't find a 'solution' folowing the thread. http://lists.iptel.org/pipermail/serdev/2005-April/004296.html Some one said that ACK/branch=0 is ok for non-negative (2xx) replies (RFC3261), but I couldn't find that text in the RFC3261. Can some one point me to the correct answer for that question? Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDIeGgiLK8unYgEMQRAsxDAJwOKxDAIHyr4Ll8hEEOIpav9FzqgACfT9Ej fSsQEexrILyfA3u7JxU6lkg= =nkWg -----END PGP SIGNATURE-----

1 0

[Serusers] SIP TEST
by fedora＠voice.dyndns.tv 09 Sep '05

09 Sep '05

Example Mailbox, There is a new voicemail in mailbox 1234: From: "Japan SIP" <8000> Length: 0:10 seconds Date: Saturday, September 10, 2005 at 12:39:17 AM Dial *98 to access your voicemail by phone. Visit http://www.voice.dyndns.tv/cgi-bin/vmail.cgi?action=login&mailbox=1234 to check your voicemail with a web browser.

1 0

RE: [Serusers] Problem with command "serctl add"
by Alberto 09 Sep '05

09 Sep '05

Sorry, I use a start-up script and the /var/log/messages : Sep 9 16:53:25 sip ser: WARNING: fix_socket_list: could not rev. resolve 192.168.100.199 Sep 9 16:53:25 sip ser: WARNING: fix_socket_list: could not rev. resolve 192.168.100.199 Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: Maxfwd module- initializing Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: new_connection(): Access denied for user 'ser'@'localhost' (using password: YES) Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: db_init(): Could not create a connection Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: ERROR: uridb_db_ver: unable to open database connection Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: ERROR: uri_db:mod_init(): Error while querying table version Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: init_mod(): Error while initializing module uri_db The script: #!/bin/sh # # Start/stop/restart SIP EXpress Router (SER) # # Version: 1.0 - Paul Belanger <pabelanger(a)gmail.com> # # Directions: # copy ser script to /usr/local/etc/rc.d/ # edit /etc/rc.conf and add the following: # ser_enable="YES" # # # 05.05.2005 - Initial Version . /etc/rc.subr name="ser" rcvar="`set_rcvar`" command="/usr/local/sbin/${name}" pidfile="/var/run/${name}.pid" load_rc_config $name ser_flags="$cron_flags -P $pidfile" run_rc_command "$1"

1 0

[Users] Call failed due to 403 Forbidden
by Subashini C V - CTD, Chennai 09 Sep '05

09 Sep '05

hi, i have registered with Asterisk, After that when i try to call my other extension, i am getting an "403 Forbidden" response. Below is the dump of SIP messages. Is there any SIP message is in conflict with the Asterisk rules..? I need your help.... >> SENT: SOURCE [0.0.0.0:1035] <-> DESTINATION [10.100.12.201:5060] INVITE sip:6442161001@10.100.12.201:5060 SIP/2.0 Via: SIP/2.0/UDP 10.100.12.209:5060;rport;branch=z9hG4bKce95dd81f969e71b577c8037f9bbdd63 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24891 INVITE Contact: "Phone1"<sip:6442091001@10.100.12.209> Content-Length: 287 Content-Type: application/sdp Expires: 180 Max-Forwards: 70 Organization: PMC-SIERRA Proxy-Authorization: DIGEST username="6442091001",realm="asterisk",nonce="51afa269",uri="sip:6442161001@ 10.100.12.201",response="4e3255ca53d7cd894c98fac21132a991",algorithm=MD5 User-Agent: Stein Authorization: DIGEST username="6442091001",realm="asterisk",nonce="51afa269",uri="sip:6442161001@ 10.100.12.201",response="4e3255ca53d7cd894c98fac21132a991",algorithm=MD5 v=0 o=UserName 584103545 0 IN IP4 10.100.12.209 s=Voice Session c=IN IP4 10.100.12.209 t=0 0 m=audio 8002 RTP/AVP 0 4 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=sendrecv a=rtcp: 8003 << RECEIVED: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6442161001@10.100.12.201> Content-Length: 0 << RECEIVED: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6442161001@10.100.12.201> Content-Length: 0 << RECEIVED: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6442161001@10.100.12.201> Content-Length: 0 >> SENT: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] ACK sip:6442161001@10.100.12.201:5060 SIP/2.0 Via: SIP/2.0/UDP 10.100.12.209:5060;rport;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 ACK Content-Length: 0 Organization: PMC-SIERRA User-Agent: Stein Thanks, Subashini DISCLAIMER This message and any attachment(s) contained here are information that is confidential, proprietary to HCL Technologies and its customers. Contents may be privileged or otherwise protected by law. The information is solely intended for the individual or the entity it is addressed to. If you are not the intended recipient of this message, you are not authorized to read, forward, print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, please notify the sender immediately by return e-mail and delete it from your computer

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