sr-users July 2006

sr-users@lists.kamailio.org

206 participants
414 discussions

[Users] Problem with avp_write
by hemant kumar 31 Jul '06

31 Jul '06

Hello List, I am using openser version: /usr/local/sbin/openserctl 1.2 - $Revision: 1.3 $ Now the problem is I used avp_write("i:45", "inv_timeout"); for the timeout and redirect the INVITE to voicemail. I got an error so i used avp_write("i:45", "$avp(s:inv_timeout)"); after doing some google searches. It seems that my change in config file didn't do any good cause now openser is giving me error 3(4772) ERROR: parse_uri: bad uri, state 0 parsed: <:hem> (4) / <:hemant@xx.xx.xx.xx> (22) 3(4772) ERROR: parse_sip_msg_uri: bad uri <:hemant@xx.xx.xx.xx> Can someone help me out here ? Thanks, Hemant

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[Users] I got an error 478,what can i do?
by feng zeheng_feng 31 Jul '06

31 Jul '06

Hi,everyone! I have installed openser 1.2 via aptitude from openser.org, and then I config openser follow the install manual( http://www.openser.org/index.php?option=com_content&task=view&id=26&Itemid=… with mysql was be installed and databases and tables was create by openser_mysql.sh. I add new user successfully like this:openserctl add fzh fzh fzh(a)myserver.com.but i can't login user fzh with window messenger or x-lite.I got an error 478---unresolvable destination(478/TM). i think there may some error in my openser.cfg. but i can't find it. i post my openser.cfg here: alias="myserver.com" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # 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(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=="myself") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.com", "subscriber")) { www_challenge("myserver.com", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; ngrep -qt port 5060 : U 2006/07/31 16:31:30.359879 59.64.183.197:5060 -> 64.233.161.83:5060 REGISTER sip:gmail.com SIP/2.0..Via: SIP/2.0/UDP 59.64.183.197;branch= z9hG4bK099e.a821a853.0.. Via: SIP/2.0/UDP 59.64.183.247:11955..Max-Forwards: 69..From: < sip:fzh@gmail.com>;tag=5ca47f5d 15c24791bb6dbfdc11aace96;epid=e655b5ac9b..To: <sip:fzh@gmail.com>..Call-ID: 135b746590ae485c93 70134a3a1400c8..CSeq: 1 REGISTER..Contact: <sip:59.64.183.247:11955>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BENOTIFY"..User-Agent: RTC/1.3.536 9 (Messenger 5.1.0639)..Event: registration..Allow-Events: presence..Content-Length: 0..P-hint : outbound.... U 2006/07/31 16:31:30.993136 59.64.183.247:1704 -> 59.64.183.197:5060 REGISTER sip:gmail.com SIP/2.0..Via: SIP/2.0/UDP 59.64.183.247:11955..Max-Forwards: 70..From: <sip:fzh@gmail.com>;tag=5ca47f5d15c24791bb6dbfdc11aace96;epid=e655b5ac9b..To: <sip:fzh@gmail.c om>..Call-ID: 135b746590ae485c9370134a3a1400c8..CSeq: 1 REGISTER..Contact: <sip:59.64.183.247: 11955>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BE NOTIFY"..User-Agent: RTC/1.3.5369 (Messenger 5.1.0639)..Event: registration..Allow-Events: pre sence..Content-Length: 0.... what can i do ? appreciate for your help fzh

1 0

[Users] Please help for TLS testing..
by Stephen Paterson 31 Jul '06

31 Jul '06

Hi Ferianto, > As I see, the error is at line 27. I see that it contain "tls_verify=1" and "tls_require_certificate=0". I don't know what is wrong with this line > because As I see from all mailinglist`s messages, they didn`t change this line and if they change it, they just change the value, for example : > tls_verify = on > tls_require_certificate = on I've just started using OpenSER today as well and had the same problem. After a little guess work I got it to work using: tls_verify_client, tls_verify_server & tls_require_client_certificate instead of tls_verify & tls_require_certificate I couldn't find this documented anywhere but it may be, I don't know my way around the docs yet. Regards Steve Steve Paterson Software Engineer Aculab Tel: +44 (0) 1908 273866 Fax: +44 (0) 1908 273801 Email: mailto:stephen.paterson@aculab.com Website: http://www.aculab.com

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[Users] Route-header DNS lookup?
by Kerker Staffan 31 Jul '06

31 Jul '06

hi i recently bounced into this problem, and i'm not sure here. i'm running the openser-devel, with the cacheless db_mode=3. (works fine btw) the record-route header received by the proxy on the other side (SNOM4S), inserts the domain name (iptel1.ipatl.se) and not the hostname (sip.iptel1.ipatl.se) in the record-route header, and uses the maddr=<ip_of_server> with the actual server IP address. now, when my client (behind the OpenSER) replies with an ACK to the incomming OK, it uses the Route-header recieved in the RR-header, and sends the ACK to OpenSER. i then get the following errors in OpenSER. --- /usr/local/sbin/openser[3583]: ERROR: mk_proxy: could not resolve hostname: "iptel1.ipatl.se" /usr/local/sbin/openser[3583]: ERROR: uri2proxy: bad host name in URI <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;maddr=172.28.248.66;transport=udp;lr> --- the ACK i sent look like this: --- Request-Line: ACK sip:2307@iptel1.ipatl.se;gruu=6fg9n6dl SIP/2.0 Via: SIP/2.0/UDP 172.28.248.52:2051;branch=z9hG4bK-d96b1fvapkyn;rport Route: <sip:172.28.248.10;lr=on;ftag=li9buf1i4p> Route: <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;maddr= 172.28.248.66;transport=udp;lr> From: "Snom 2652" <sip:2652@ipatl.se>;tag=li9buf1i4p To: <sip:2307@ipatl.se>;tag=hvseiz7kgb Call-ID: 3c269d83900b-xj3ild14y880@snom360 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:2652@172.28.248.52:2051;line=cp4a7ljd> Content-Length: 0 --- as far as i understand, according the rfc 3263, the route-header may contain domain name that has to be resolved using SRV. --- "6 Constructing SIP URIs In many cases, an element needs to construct a SIP URI for inclusion in a Contact header in a REGISTER, or in a Record-Route header in an INVITE. According to RFC 3261 [1], these URIs have to have the property that they resolve to the specific element that inserted them. However, if they are constructed with just an IP address, for example: sip:1.2.3.4 then should the element fail, there is no way to route the request or response through a backup. SRV provides a way to fix this. Instead of using an IP address, a domain name that resolves to an SRV record can be used: sip:server23.provider.com" --- now, OpenSER only asks DNS for an A record of the name recieved in the route header, and since that's a domain name, it's unresolvable, and so the ACK is never sent. any hints or clues? best regards, /Staffan Kerker -- Staffan Kerker Saab Communications, Växjö p. +46 470 42185 c. +46 705 391365 m. staffan.kerker(a)saabgroup.com

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[Users] call management software
by ram 31 Jul '06

31 Jul '06

Hi all Is any one here in the group used any call management software for SER I have seen software for Asterisk hudlite, does any kind of software for SER any one tested and report their experiences why iam asking is after Ser running some days i get this error in my syslog sl_reply_error used: I'm terribly sorry, server error occured (2/SL) that time my users not able to make call, i keep get fast busy, that time i need to reboot the Ser so calls again start going out side If any kind of monitoring software available or do automated job, when ever found that calls are not going then restart SER or some kind of solution should help the admin to chek the logs, when the SER restarted. If not some one need to monitor constantly or wait for customer to call to support \the calls are not going then we restart service will back the services, but i feel bad practice, before customer realise some problem with server, the Admin able to fix the problem any suggestions will be appriciated. ram,

2 1

[Users] Openser 1.0.1 installation problems
by J K 31 Jul '06

31 Jul '06

Hi there. I'm trying to install Openser 1.0.1 on CentOS 4.3. I've untarred the source code using 'tar zxvf openser-1.0.1_src.tar.gz' to a directory openser-1.0.1, but when I type 'make all' or 'make prefix=/ all' to build, I get the following errors: make: Warning: File `Makefile.rules' has modification time 1.4e+07 s in the future bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex make: Warning: File `Makefile.rules' has modification time 1.4e+07 s in the future bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex make: *** Deleting file `sr_module.d' make: *** [sr_module.d] Interrupt Anyone have any idea how I can fix this? Thanks in advance. __________________________________________ BreakThru.com - Pure email. Zero spam. http://friendfinder.com/go/f47930 - Dating Personals and Social Networking http://quizhappy.com/go/f301 - Ph.D certified tests and quizzes http://dine.com/go/f5554 - Restaurant Reviews in your area

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[Users] Resetting flags in branch_route[] resets flags for ALL branches
by Tavis P 31 Jul '06

31 Jul '06

I'm using branch route to trigger mediaproxy logic per branch for UAC directed calls and i'm experiencing some strange behavior: t_relay() is called, there are two branches In both branches there are two flags set During branch_route processing of the first branch both are left intact. During branch_route processing of the second branch, both are reset (unset) The destination UAC for the first branch picks up the phone call, sending a SIP OK message This is where i'm experiencing the problem Inside of the onreply_route both flags are NOT set, even though they were unset in a different call branch Has anyone experienced this behavior before? Is it supposed to work this way? This is using a recent (1-2 weeks ago) cvs checkout of OpenSER 1.0.0 tavis

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[Serusers] How to handle redirect with sipbroker?
by Roger Lewau 30 Jul '06

30 Jul '06

I dont know if any of you use the SIPBroker service to do multi enum lookups, but I'm hoping there are some on this list who do. I have been using SIPBroker for dialing with sip codes only so far and did my enum lookups on my own. But now, I decided to set ser up to try sipbroker first for all external calls and then fall back to my different gateways if no route can be found via SIPBroker. SIPBroker, in prinicpal, works like this. I send an Invite for any enum number to SIPBroker. If sipbroker finds a route for that number to another voipprovider it will proxy the call to the found provider. If sipbroker can not find a route, it will reply with a redirect of the call to my self and I am supposed to handle the call setup my self, through my gateways. My question is how to handle this redirect message? Any one who has a working failure route to handle this situation, and are willing to share? Here is an actual SIP conversation, initiated from Asterisk via SER, of a failed Enum lookup through SIPBroker: # U 212.247.91.XXX:5060 -> 24.196.79.163:5060 INVITE sip:4640240252@sipbroker.com SIP/2.0. Record-Route: <sip:212.247.91.XXX;ftag=as5e811304;lr=on>. Via: SIP/2.0/UDP 212.247.91.XXX;branch=z9hG4bK7307.c7d88007.0. Via: SIP/2.0/UDP 212.247.91.XXZ:5060;branch=z9hG4bK05f7cabc;rport=5060. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. To: <sip:240252@sip.serverhallen.com>. Contact: <sip:330000@212.247.91.XXZ>. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Date: Sat, 29 Jul 2006 22:42:23 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 336. . v=0. o=root 850 851 IN IP4 212.247.91.XXZ. s=session. c=IN IP4 212.247.91.XXZ. t=0 0. m=audio 34852 RTP/AVP 0 8 18 97 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:97 iLBC/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. ## U 24.196.79.163:5060 -> 212.247.91.XXX:5060 SIP/2.0 300 Redirect. Via: SIP/2.0/UDP 212.247.91.XXX;branch=z9hG4bK7307.c7d88007.0. Via: SIP/2.0/UDP 212.247.91.XXZ:5060;branch=z9hG4bK05f7cabc;rport=5060. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. To: <sip:240252@sip.serverhallen.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe4c. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. CSeq: 103 INVITE. Contact: sip:4640240252@sip.serverhallen.com. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 24.196.79.163:5060 "Noisy feedback tells: pid=15326 req_src_ip=212.247.91.XXXreq_src_port=5060 in_uri=sip:4640240252@sipbroker.com out_uri=sip:4640240252@sip.serverhallen.com via_cnt==2". . # U 212.247.91.XXX:5060 -> 24.196.79.163:5060 ACK sip:4640240252@sipbroker.com SIP/2.0. Via: SIP/2.0/UDP 212.247.91.237;branch=z9hG4bK7307.c7d88007.0. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. To: <sip:240252@sip.serverhallen.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe4c. CSeq: 103 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/freebsd)). Content-Length: 0. .

1 0

[Serusers] Mediaproxy accounting - SER
by Robert Zorop 29 Jul '06

29 Jul '06

Hi, is there anyone experimenting problems with dbaccounting with mediaproxy ?. I have installed 1.7.2 and it works in some situations and doesnt in others. The calls are terminating in a cisco 2600 equip. The log i have is : delete 347395089(a)192.168.0.102 info= session 347395089(a)192.168.0.102: 93/3917/4010 packets, 17736/782484/800220 bytes (caller/called/relayed) session 347395089(a)192.168.0.102: ended. But no account is done. Sincerely i don't really know what is happening here. Please any help/clue provided will be a great help .

1 0

[Serusers] Implementing Group Pickup
by Simon 29 Jul '06

29 Jul '06

Hi, Has anyone thought how to implement group pickup purely in SER ? Currently we are using Asterisk for all PBX type functions, but want to move to a design where SER handles the dialogue with the sip phones and uses Asterisk purely as a voice mail server. So has anyone implemented group pickup and hunt groups before ? Many Thanks Simon

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sr-users July 2006