- sr-users - kamailio.org

[OpenSER-Users] about openSER and SIPp
by bozhang＠nec-labs.com 02 Aug '07

02 Aug '07

Hi there, I'm doing sipp performance measurements with openser 1.2. I run OpenSER on machine A, then run one SIPp on machine B, one SIPp on machine C. For the SIPp on machine C, I start it using UAS mode, ie., ./sipp -sn uas For the SIPp on machine B, if I start is using the following command: ./sipp -sn uac C -rsa A -m 1 -r 1 Everything is correct. The OpenSER can relay messages correctly. But if I start SIPp on machine B using the following command: ./sipp -sn uac -rsa A -m 1 -r 1 -d 5000 Then the SIPp client on machine B will crash because of receiving unexpected "ringing 180" message. Note: the only difference is I add a pause 5 seconds using "-d 5000" option. What I found is: the OpenSER server will keep sending "INVITE" to C during the '5 second pause', so C will keep reponding with "180 Ringing" to B. Does anybody know why OpenSER keeps sending "INVITE" to C while B only issues one "INVITE" request? BTW, the configuration file for OpenSER is very simple: # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=32 log_facility=LOG_LOCAL0 disable_tcp=yes disable_dns_blacklist=yes disable_dns_failover=yes # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # listen=udp:192.168.2.102:5060 # ------------------ module loading ---------------------------------- loadmodule "modules/tm/tm.so" modparam("tm", "wt_timer", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ t_relay(); } Thanks a lot! -Bo Zhang

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Re: [OpenSER-Users] Dispatcher
by Daniel-Constantin Mierla 02 Aug '07

02 Aug '07

OK, so you didn't get in the condition of the failure route. Daniel On 08/02/07 12:58, Thomas Deillon wrote: > Hello, > > It's ok, I finally find the solution with raviprakash sunkara. > It was the parameter of tm. fw_timer. > > Cheers, > > Thomas > > On 02/08/07, *Daniel-Constantin Mierla * <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > can you run in debug=7 and log_stderror=yes and watch the debug > messages? If you have issues understanding what is happening, then > just > forward it to me along with dispatcher.list file. > > Daniel > > On 08/01/07 15:56, Thomas Deillon wrote: > > Hi, > > > > I'm using OpenSER as PROXY and REGISTRAR server, a Cisco as Gateway > > and 2 Asterisks (using dispatcher) as Voicemail and conference > Servers. > > > > It works perfectly on load-balancing, but not in failover because I > > never go on the failure route. > > Do you have a idea (a part of my config file below) ? > > rq: Just for information, the 12 is the milliwatt() function on > Asterisks. > > > > my openser.cfg > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > modparam("usrloc", "db_url", "mysql://openser:openser@localhost > /openser") > > modparam("usrloc", "db_mode", 1) > > > > > modparam("dispatcher","list_file","/usr/local/etc/openser/dispatcher.list") > > modparam("dispatcher","flags",2) > > modparam("dispatcher", "dst_avp", "$avp(i:271)") > > modparam("dispatcher", "grp_avp", "$avp(i:272)") > > modparam("dispatcher", "cnt_avp", "$avp(i:273)") > > modparam("dispatcher", "force_dst", 1) > > > > route { > > [...] > > > > if (method=="INVITE") > > { > > if (uri=~"sip:12@.*") > > { > > xlog( "L_ERR", "A: $rm $fu:$op -> $du AVP: $avp(i:271), > > $avp(i:272), $avp(i:273)\n" ); > > ds_select_dst("1", "4"); > > #t_relay(); > > xlog( "L_ERR", "B: $rm $fu:$op -> $du AVP: $avp(i:271), > > $avp(i:272), $avp(i:273)\n" ); > > t_on_failure("1"); > > } > > } > > [...] > > } > > > > failure_route[1] > > { > > xlog( "L_ERR", "C: $rm $fu:$op -> $du\n" ); > > if(t_check_status("408")) > > { > > ds_mark_dst(); > > ds_next_dst(); > > } > > else { t_reply("501", "Not Implemented"); } > > > > xlog( "L_ERR", "D: $rm $fu:$op -> $du\n" ); > > } > > > > Cheers, > > > > Thomas > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

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[OpenSER-Users] Dispatcher
by Thomas Deillon 02 Aug '07

02 Aug '07

Hi, I'm using OpenSER as PROXY and REGISTRAR server, a Cisco as Gateway and 2 Asterisks (using dispatcher) as Voicemail and conference Servers. It works perfectly on load-balancing, but not in failover because I never go on the failure route. Do you have a idea (a part of my config file below) ? rq: Just for information, the 12 is the milliwatt() function on Asterisks. my openser.cfg modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_url", "mysql://openser:openser@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("dispatcher","list_file","/usr/local/etc/openser/dispatcher.list") modparam("dispatcher","flags",2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "force_dst", 1) route { [...] if (method=="INVITE") { if (uri=~"sip:12@.*") { xlog( "L_ERR", "A: $rm $fu:$op -> $du AVP: $avp(i:271), $avp(i:272), $avp(i:273)\n" ); ds_select_dst("1", "4"); #t_relay(); xlog( "L_ERR", "B: $rm $fu:$op -> $du AVP: $avp(i:271), $avp(i:272), $avp(i:273)\n" ); t_on_failure("1"); } } [...] } failure_route[1] { xlog( "L_ERR", "C: $rm $fu:$op -> $du\n" ); if(t_check_status("408")) { ds_mark_dst(); ds_next_dst(); } else { t_reply("501", "Not Implemented"); } xlog( "L_ERR", "D: $rm $fu:$op -> $du\n" ); } Cheers, Thomas

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[OpenSER-Users] Header Mangling
by Peter Gradwell 02 Aug '07

02 Aug '07

hi ya, When we send phone call out to our PSTN gateways and carriers, we often have to modify the To and From headers to put them in the right syntax and to copy the caller id up from the remote-party-id header. We think the uac module will allow us to do this on the From header, but we also need to add/replace digits in the To header, e.g. changing it into INTL format, from 01225 to 441225... If we do this using TextOps then the problem is that it only changes the string sent to the Telco switch, and it is not therefore changed on the reply, and so the end user agent does not then handle the replies properly. Does anyone know how we can mangle the To header? Is there a similar module to uac for To headers? cheers peter

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[OpenSER-Users] Attribue User-name was not found in request, unique ID MAY be inconsistent
by zze-WU Yin RD-MAPS-LAN 02 Aug '07

02 Aug '07

Hi, everybody Another probleme with Openser and Freeradius. When I tried to use the acc module, i recieved: rlm_acct_unique: WARNING Attribue User-name was not found in request, unique ID MAY be inconsistent And in the Accounting-Request packet, i can not find User-name. Do anybody know why? Thank you very much!!

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[OpenSER-Users] Pb with log_fmt
by zze-WU Yin RD-MAPS-LAN 02 Aug '07

02 Aug '07

Hi, thank you very much. I know what the problem is now. Because I use a openser-1.2.1-notls. So the "log_fmt" is obsolete. In the http://www.openser.org/docs/modules/1.2.x/acc.htm it wrote that: The accouting module will log by default a fixed set of attributes for the transaction - if you customize you accouting by adding more information to be logged, please see the next chapter about extra accouting - Section 1.2. The fixed minimal accouting information is: Request Method name >From header TAG parameter To header TAG parameter Call-Id 3-digit Status code from final reply Reason phrase from final reply Time stamp when transaction was completed If a value is not present in request, the empty string is accounted instead. I want have a User-Name in the accounting information. So I saw in the Section 1.2 Selection of extra information is done via xxx_extra parameters by specifying the names of additional information you want to log. This information is defined via pseudo-variables and may include headers or AVPs values or other message or system values. The syntax of the parameter is: xxx_extra = extra_definition (';'extra_definition)* extra_definition = log_name '=' pseudo_variable And the pseudo_variable I need is '$au' for User-Name. But I don't understand how to use this paramater in the openser.cfg. Do you have some ideas??

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[OpenSER-Users] Fw: Adding an attribute to SDP
by Kuddusi 01 Aug '07

01 Aug '07

Hi, How can I add a header to SDP? Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default line to SDP body. Like if (has_body()){ if (!search_body("ptime") { # like below à # append_body("a=ptime:20"); } } Thanks, Kuddusi CIFTCIBASI Next Generation Networks Teletek Telecommunication Services Corp. Ayazma Dere Cad. Aksit Plaza No:12/1 Fulya, Besiktas , 34349, www.teletek.net Tel : +90 212 227 7030 Direkt: +90 212 310 2233 Fax : +90 212 227 8700 Email: kuddusi.ciftcibasi(a)teletek.net Skype : realkudu MSN : kuddusic(a)hotmail.com Yahoo: kuddusi.

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[OpenSER-Users] Adding a header to SDP
by Kuddusi CIFTCIBASI 01 Aug '07

01 Aug '07

Hi, How can I add a header to SDP? Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default header to SDP body. Like if (has_body()){ if (search_body("ptime") { # like below --> # append_body("a=ptime:20"); } } Thanks, Kuddusi CIFTCIBASI Next Generation Networks Teletek Telecommunication Services Corp. Ayazma Dere Cad. Akşit Plaza No:12/1 Fulya, Beşiktaş , 34349, www.teletek.net Tel : +90 212 227 7030 Direkt: +90 212 310 2233 Fax : +90 212 227 8700 Email: kuddusi.ciftcibasi(a)teletek.net Skype : realkudu MSN : kuddusic(a)hotmail.com <mailto:kuddusic@hotmail.com> Yahoo: kuddusi. Yasal Uyari: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Teletek Telekomünikasyon Hizmetleri A.Ş. onayi olmaksizin içerigi kopyalanamaz, üçüncü kişilere açiklanamaz veya iletilemez . Bu mesajin gönderilmek istendigi kişi degilseniz (ya da bu e-posta'yi yanlişlikla aldiysaniz), lütfen yollayan kişiyi haberdar ediniz ve mesaji sisteminizden derhal siliniz. Teletek bu mesajin içerdigi bilgilerin dogrulugu veya eksiksiz oldugu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriginden, iletilmesinden, alinmasindan, saklanmasindan sorumlu degildir. Bu mesajin içerigi yazarina ait olup, Teletek'in görüşlerini içermeyebilir.Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without Teletek Telekomünikasyon Hizmetleri A.Ş.'s authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Teletek makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Teletek.

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[Serusers] xcap module compilation issues
by Parminder.Nayyar＠relianceada.com 01 Aug '07

01 Aug '07

Hi Can anyone help me out in compiling xcap module?? as far as the module documentation concerns... its says.. that libcurl.so should be there already.. but how do i get that i downloaded a curl.tar.gz from internet.. gunzip n untar it.... and then compiled it but can't find any .so made out of it more so, the xcap module when compiled.. gives error that it can't find curl/curl.h What do i do?? Can any one help??? Parminder Nayyar

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[OpenSER-Users] Problem with "log_fmt"
by zze-WU Yin RD-MAPS-LAN 01 Aug '07

01 Aug '07

Hi, everybody I use openser and want to use acc module. But when I tryed to use modaparm("acc", "log_fmt", "mfs") openser can not start. Can you tell me why?

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[Serusers] How to register call forward(busy, no answer, all) from the UserAgent?
by Sean 01 Aug '07

01 Aug '07

Greetings, I would like to register the call forward functions for my UA, is it available for registering these functions from UserAgent? for example, via the REGISTER sip message or other messages? can anyone help me for this topic? Thanks a lot. Sean ____________________________________________________________________________________ Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/

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[OpenSER-Users] Can't use variables as function parameter?
by Iñaki Baz Castillo 01 Aug '07

01 Aug '07

Hi, in URI as "ping(a)domain.org" I need to change URI to just "domain.org" (without the username@). But I have multidomain so can't rewrite uri directly and must use the requested URI domain ($rd). So I try: if (uri=~"sip:ping[@]+.*") { rewriteuri("sip:$rd"); xlog("L_INFO", "-- New URI = $ru\r\n"); ... but I get: -- New URI = sip:$rd so it seems that I cannot use variables for making function parameters, is it true? so then, how should I do it? maybe I need using AVP's for that? Thanks for all. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Info about "TM/473 Filtered"?
by Iñaki Baz Castillo 31 Jul '07

31 Jul '07

Hi, I'd like to understand how TM module filters "sometimes" some calls. For now, I just think it there isn't response and the call is repeated sometimes then TM module filteres the following calls. I find no info about how it works in the doc, could somebody give me a link or so to a document with the explanation of this issue? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Colored logs in a file
by Iñaki Baz Castillo 31 Jul '07

31 Jul '07

Hi, I use colors for the XLOGs with log_stderror=yes, this is nice when I'm learning and debuggoing all the time. But for a production system I know I need to disable log_stderror and log into Syslog (for example using a specific local facility for logging in an independent file /var/log/openser.log). But in that way I loose my cool colors that help me a lot debugging. A solution I'm thinking about is to use "log_stderror=yes" and start OpenSer redirecting the output to a file, but sincerely I don't think this is very effective and serious. Any suggestion? maybe I should forget my funny colors? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Privider SIP/PSTN
by Marc LEURENT 31 Jul '07

31 Jul '07

I have a SIP/PSTN provider with an account username/password. The address of the provider is sip.test.com I would like to redirect phone calls to the gateway, but how is it possible to log in on the gateway and forward calls? Thank you for your replies Best Regards, I've tried: if (uri=~"0[0-9][0-9]*@*") { rewritehostport("username:password@sip.test.com"); # Rewrite the URI route(1); exit; }

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[OpenSER-Users] Parse error in "avpops" module
by Iñaki Baz Castillo 31 Jul '07

31 Jul '07

Hi, using OpenSer 1.2.1 and trying to use "usr_preferences" MySQL table for loading AVP's. My conf: loadmodule "avpops.so" ... modparam("avpops", "db_url","mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table","usr_preferences") modparam("avpops","use_domain",1) ... route[5] # Auth { line #543: if (avp_db_load("$ru/domain","s:trusted_ips") && avp_check("s:trusted_ips", "eq/$sp/gi")) { xlog("L_INFO", "Authorized by usr_preferences AVPs\r\n"); } .... } I get this error: 0(0) ERROR:avops:parse_avp_db:: error - bad avp flags 0(0) ERROR:avpops:fixup_db_avp: parse failed 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 543 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration Any idea of what I'm doing wrong? Thanks a lot for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] nathelper without rtpproxy
by Nhadie Ramos 31 Jul '07

31 Jul '07

Hi All, Has anyone tried nathelper but not using rtpproxy, my server is on a location with limited bandwidth so i'd like to be able to just have the sip messages to the server but not the media. Do i need all phones configured to use STUN if the SIP server is configured that way? Thank You in Advanced Regards, Nhadie

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[Serusers] nathelper without rtpproxy
by Nhadie Ramos 31 Jul '07

31 Jul '07

Hi All, Has anyone tried nathelper but not using rtpproxy, my server is on a location with limited bandwidth so i'd like to be able to just have the sip messages to the server but not the media. Do i need all phones configured to use STUN if the SIP server is configured that way? Thank You in Advanced Regards, Nhadie

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[OpenSER-Users] OpenSER + Freeradius + CDRTool
by Marc LEURENT 30 Jul '07

30 Jul '07

Hello, I'm still trying to get openser + freeradius + cdrtool working together I have stil one problem, it's that UserName are not stored in the acct database, whereas they are displayed in the aact logs Any piece of advice will be welcome Thanks This is a view of the CDRTool interface: SIP Caller, In, SIP destination, don't appear in the database! Id Start time SIP Proxy SIP caller In SIP destination Out Dur Price KBIn KBOut Status Codec 3N 2007-07-30 10:00:19 05:22 0.00 0.00 Ok (200) Signalling information Session: 1ec37-c0a80101-0-2(a)192.168.95.4 Start time: 2007-07-30 10:00:19 Europe/Paris Stop time: 2007-07-30 10:05:41 Europe/Paris Method: INVITE from : From: Dialed URI: Next hop URI: Billing Party: This is a part of the acct log Mon Jul 30 13:55:19 2007 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = INVITE Event-Timestamp = "Jul 30 2007 13:55:19 CEST" Sip-From-Tag = "c0a80101-6e0646" Sip-To-Tag = "c0a80101-888e30" Acct-Session-Id = "3f9aaaf-c0a80101-0-10(a)192.168.95.4" Sip-Src-IP = "192.168.95.4" Sip-Src-Port = "5060" User-Agent = "THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-AE" User-Name = "105" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.95.248 Client-IP-Address = 192.168.95.248 Acct-Unique-Session-Id = "694a4cf65dd7fd9f" Timestamp = 1185796519

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[Serusers] Bad Config error in SER.CFG when tried with SEMS
by Ranganath B 30 Jul '07

30 Jul '07

Hi, When I was trying to bringup SER with SEMS installed I get the following error in /var/log/messages: Jul 30 16:24:21 Emailclient ser: parse error (214,17-18): parse error Jul 30 16:24:21 Emailclient ser: parse error (214,19-20): bad arguments The line number 214 refers to "load_avp("ruri","email",0);" Please provide me some input on getting rid of this issue. Thanks in advance, Ranganath B

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[OpenSER-Users] Mediaproxy performance
by Tung Tran 30 Jul '07

30 Jul '07

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[Serusers] RTP capacity
by Tung Tran 29 Jul '07

29 Jul '07

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[Serusers] consultant needed
by Theodore Cekan 29 Jul '07

29 Jul '07

Hello, We are looking to hire a consultant to help us design and configure Ser for a residential service offering. We are based in Denver Colorado. If you know of anyone who could help please let me know. Ted Cekan 303-692-9409 x304

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[Users] provisional_reply_timer
by Klaus Darilion 28 Jul '07

28 Jul '07

Hi! i wonder if there is a timer which controls the timeout for provisional replies. E.g. I want to cancel an INVITE and retry another gateway if the gateway does not respond with 100 (or 180) in a certain time. regards klaus -- Klaus Darilion nic.at

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[OpenSER-Users] Unable to configure LCR in openser - problem with next_gw()?
by Miodrag Radulovic 28 Jul '07

28 Jul '07

Hello everyone! Please help... I've started testing openser 1.2.1 few weeks ago (on freebsd 6.2, I've got it from freebsd port collection) and I simple can not make LCR work at all. I spent last few hours reading almost all the posts on this mailing list and I can see it is working but I can not figure it out why mine configuration is not... For simple start - I want all calls where destination number starts with 9 to be routed to PSTN GW. So I made these entries in LCR tables: [root@santos /var/log]# openserctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 10 | 9 | % | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateway groups +--------+----------+ | grp_id | grp_name | +--------+----------+ | 1 | comutel | +--------+----------+ lcr gateways +---------+---------------+------+------------+-----------+--------+-------+--------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | prefix | +---------+---------------+------+------------+-----------+--------+-------+--------+ | c2811 | 192.168.253.1 | 5060 | 1 | 1 | 1 | 0 | | +---------+---------------+------+------------+-----------+--------+-------+--------+ For openser.cfg itself I've started with sip:wizard configuration template (SIP Proxy/Registrar with Offnet-Termination and Accounting) then I made some necessary changes (I do not have Mediaproxy or RTPproxy and added domain and populated trusted table). But when I dial 93016576 I always have this error message in xlog output: Jul 19 18:50:24 santos /usr/local/sbin/openser[1412]: No PSTN gateways available - M=INVITE RURI=sip:93016576@192.168.253.205:5060 F=sip:lvs1@192.168.253.205:5060 T=sip:93016576@192.168.253.205:5060 IP=192.168.253.202 ID=676c7a05-4bcd2745(a)192.168.253.202 Here is part of openser.cfg where next_gw() fails: route[10] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); exit; } route(12); if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(6); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); sl_send_reply("404", "User Not Found"); exit; } I believe that I've made some errors in LCR routes entries, but documentation is really sparse on this issue... Please note - I've tried replacing "%" with empty string in from_uri column in routes table but then openser was not able to startup: [root@santos /var/log]# openserctl lcr rmroute 9 % 1 1 MySql password for user 'openser@localhost': 200 OK [root@santos /var/log]# openserctl lcr addroute 9 '' 1 1 MySql password for user 'openser@localhost': 200 OK [root@santos /var/log]# openserctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 11 | 9 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateway groups +--------+----------+ | grp_id | grp_name | +--------+----------+ | 1 | comutel | +--------+----------+ lcr gateways +---------+---------------+------+------------+-----------+--------+-------+--------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | prefix | +---------+---------------+------+------------+-----------+--------+-------+--------+ | c2811 | 192.168.253.1 | 5060 | 1 | 1 | 1 | 0 | | +---------+---------------+------+------------+-----------+--------+-------+--------+ [root@santos /var/log]# openserctl restart Stopping OpenSER : \E[37;33mINFO: stopped Starting OpenSER : \E[37;31mERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed And here is relevant part from openser log output: Jul 21 21:30:26 santos /usr/local/sbin/openser[10216]: ERROR:lcr:load_regex: bad from_uri re Jul 21 21:30:26 santos /usr/local/sbin/openser[10216]: ERROR: lcr:mod_init(): failed to reload gateways and routes Jul 21 21:30:26 santos /usr/local/sbin/openser[10216]: init_mod(): Error while initializing module lcr Please can someone help and point me in the right direction? BR, Miodrag

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