sr-users December 2006

sr-users@lists.kamailio.org

152 participants
258 discussions

[Serusers] lookup() and forking
by O'Shaughnessy Evans 11 Dec '06

11 Dec '06

Hello. I'm trying to debug a problem I seem to be having with parallel forking and I'd like to check my understanding of ser's behavior on a few points. I have 2 phone numbers registered in the location table for a certain subscriber, each with a q value of 1. When I call the subscriber number, I don't see ser forking and then sending an INVITE to each contact, but only the one with the lower expiration value (the first number that "serctl ul show" lists). I have append_branches set to 1 and desc_time_order set to 0. The way I've understood everything I've read, lookup() should be laying the groundwork for the tm module to fork ser so that when I call t_relay ser will intiate one INVITE to each location with the same q value. Is this not correct? Is there anything special I should be doing to tell it to fork an INVITE to each contact at this point? Thanks, and best regards. -- = o'shaughnessy_evans -=- sys_admin @ (aloha|ilhawaii|myworld|turquoise).net = Pacific LightNet, Inc. "Beware the lollipop of mediocrity. Once you lick it you will suck forever!" -- Capt. Gladys Stotephamphlet

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[Serusers] Subscriber table columns phplib_id, domn & uuid usage
by Kamal.Mann＠t-systems.com 11 Dec '06

11 Dec '06

Hi All I need to know the usage of 'phplib_id, domn & uudi' columns into ser database table subscriber. Please give me your valuable suggestion for the same. Thanks in anticipation Kamal Mann

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[Serusers] chain of rtp proxies ...
by Cesc 11 Dec '06

11 Dec '06

Hi, I am currently testing a system which connects two "islands" ... each island has its own SER (0.9.6) + rtp proxy ... Actually there is NOT going to be NAT or firewall among them, but i still do want to proxy all RTP streams (sorry ... requirements). Probably even within the phones in the same island ... I have a few questions ... - i saw mentioned that chaining is possible ... no problem there right? i need to send an extra parameter to the force_rtp_proxy and that is it? no side-effects if, i.e, call between the phones in the same island (thus, just one rtp proxy)? - What would be the bare minimum config to force rtpproxy, without all the NAT tests? as said, i have no such problem, so i would like a minimal config: fix_nated_sdp and force_rtp_proxy? is that enough? - For testing purposes in my little lab ... can i run two SERs on the same box (that i know i can do :D ) ... which use the same rtpproxy on the box, thus via the same unix socket? just for testing ... they put me in a corner of an office with just a little table :) Regards, Cesc

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[Serusers] Error while loading TM module. Please help!!
by Meidiana Ten 11 Dec '06

11 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") 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", "/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", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") ----- ----- -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

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[Users] How to communicated Both SIP and IAX2 each other ?
by raviprakash sunkara 11 Dec '06

11 Dec '06

Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)----------->OpenSER------> Asterisk------> UA(IAX2) . UA(IAX2) ------- >Asterisk --- > OpenSER ------ > UA (SIP ). other wise we can like that.. UA(SIP ) ----------- > Asterisk--------->UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] Mediant 2000 and SER, caller ID issue
by Nick De Cristofaro 11 Dec '06

11 Dec '06

Hello all, Trying to debug a problem in regards to Caller IDs not being showed through our PSTN gateway. Not sure if this is an issue with the ser or with the mediant 2000 gateway or somewhere else but any help to this issue would be great. I am new to this and we are trying to solve this problem, basically one of our boxes isn't providing caller ID to the PSTN network when users are calling. Basically they will show up as "private number" when calls are initiated from inside and terminating on the PSTN networking, when they are terminating inside the ser itself the caller ID shows up with no problem, but not on the outside. We wish to have the ability to show private numbers and to show public numbers, but as far as it goes right now everyone is supposed to have public numbers. Where might the problem be? Any help would be great Thanks Nick

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Re: [Serusers] chain of rtp proxies ...
by Jiri Kuthan 11 Dec '06

11 Dec '06

At 18:46 09/12/2006, Cesc wrote: >A final question ... basically thinking out loud (and writing it down) :) >I read that rtpproxy won't start relaying until it got an rtp packet >from both sides ... is it true? yes >could this not cause problems, >specially with chained rtpproxies, if say, i have one of the phones >not sending rtp packets (say, it starts muted ... muted means no rtp >packets)? indeed but that's it. MS Windows Messenger does that for example with voice inactivity detection. You have to speak to hear. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] mon script for openser ?
by G.Jacobsen 10 Dec '06

10 Dec '06

Hello, I intend to test openser health periodically with a (perl) script to see whether it is answering SIP requests - to be used for mon in combination with heartbeat. If anyone has done such thing I would be grateful for any code snippets. I suppose as another possibility one could use one of the Nagios sipsak scripts and modify that for mon ? Gerry

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Re: [Serusers] account forwarded calls
by Mobashir Ahmed 10 Dec '06

10 Dec '06

this has always been a problem with Snom Sip Proxy. we are looking for a solution on this past many months. Please hlep.!! Regards, Mobashir Ahmed. ----- Original Message ---- From: Rosario Pingaro <rpingar(a)nesec.it> To: serusers(a)lists.iptel.org Sent: Sunday, December 10, 2006 2:46:20 PM Subject: [Serusers] account forwarded calls I implememented call forwarding using the exemple found on the onsip.org website. It is working fine but I have an accounting problem. Infact the forwared call is not accounted at all. This means that the accounting modle records the call as it is in the beginning ant not the second leg. Is there a way tha have accounted the forwarded call as originated from the forwader to bill him? Regards Rosario _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com

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[Serusers] account forwarded calls
by Rosario Pingaro 10 Dec '06

10 Dec '06

I implememented call forwarding using the exemple found on the onsip.org website. It is working fine but I have an accounting problem. Infact the forwared call is not accounted at all. This means that the accounting modle records the call as it is in the beginning ant not the second leg. Is there a way tha have accounted the forwarded call as originated from the forwader to bill him? Regards Rosario

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sr-users December 2006