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37470 discussions

[Kamailio-Users] LDAP Module and TLS/SSL
by Sven Schulz 15 Mar '09

15 Mar '09

HI, Ive tested the LDAP authentication using Kamailio 1.4.2 and it works great. However, I'd like prevent the passwords being sent back "in the clear" from the LDAP server. Is there a way to use the Kamailio ldap module with TLS/SSL? I already have openssl and openldap-client on the kamailio box and can successfully use the ldapsearch command line tool via TLS. Sven Schulz Penn State University

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[Kamailio-Users] About "476 Unresolvable destination"
by Iñaki Baz Castillo 14 Mar '09

14 Mar '09

Hi, "476 Unresolvable destination" is returned by Kamailio when the destination domain has no resolution. Is it a custom reply code? is it part of some RFC/draft? Thanks. -- Iñaki Baz Castillo

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[Kamailio-Users] ERROR:error opening Openser's FIFO FIFO
by Metshet Seleshi 13 Mar '09

13 Mar '09

please help , why do i get ERROR:error opening Openser's FIFO FIFO ERROR:make sure you have the line 'Modparam('mi_fifo', 'fifo_name', 'FIFO' in your config---which I have ERROR:and also have loaded the mi_fifo module.- yes, it is loaded, I cant do openserctl moni, any idea?? regards ________________________________ From: "users-request(a)lists.kamailio.org" <users-request(a)lists.kamailio.org> To: users(a)lists.kamailio.org Sent: Thursday, March 12, 2009 2:36:05 PM Subject: Users Digest, Vol 46, Issue 39 Send Users mailing list submissions to users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.kamailio.org You can reach the person managing the list at users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: kamailio start with error Access denied for user.. (svenp) 2. Using fr_inv_timer_avp (mayamatakeshi) 3. DID assignment & routing (Robert McGilvray) 4. Does Kamailio can be used as ptt server? (doolin wu) 5. Siremis: login problems (Luntras Loredana) 6. Re: Siremis: login problems (Luntras Loredana) ---------------------------------------------------------------------- Message: 1 Date: Thu, 12 Mar 2009 04:44:16 -0700 (PDT) From: svenp <s.pastorik(a)westfalen-blatt.de> Subject: Re: [Kamailio-Users] kamailio start with error Access denied for user.. To: users(a)lists.kamailio.org Message-ID: <22474624.post(a)talk.nabble.com> Content-Type: text/plain; charset=us-ascii Hello, i foud the error in my config. I think user and pw is in config file kamailio.cfg an alias from file kamctlrc. original: modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") The right Username und PW must configured in the kamailio.cfg too. My kamailio Server is now running :). -- View this message in context: http://www.nabble.com/kamailio-start-with-error-Access-denied-for-user..-tp… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. ------------------------------ Message: 2 Date: Thu, 12 Mar 2009 20:54:49 +0900 From: mayamatakeshi <mayamatakeshi(a)gmail.com> Subject: [Kamailio-Users] Using fr_inv_timer_avp To: kamailio <users(a)lists.kamailio.org> Message-ID: <15b9404e0903120454s34c0b9eqd57bf5313d89ba75(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Hello, I'm seeing the following issues when using the latest kamailio 1.4 (Revision: 5689) to set an AVP to control fr_inv_timer: - if I set the AVP to an empty string, according to tm doc, the value of the timer would fall back to the default value. This doesn't happen and the fr_inv_timer is set to zero (not a problem as I just have to set it to the static value. It is just inconsistent with the doc). - if I try to set the timer to something bigger than 60 seconds, the setting doesn't work and it seems the default value is used instead (I tested this with both hard-coding the assignment in the cfg and loading the avp from the usr_preferences). regards, takeshi ------------------------------ Message: 3 Date: Thu, 12 Mar 2009 10:01:40 -0400 From: "Robert McGilvray" <rmcgilvr(a)globeop.com> Subject: [Kamailio-Users] DID assignment & routing To: <users(a)lists.kamailio.org> Message-ID: <888FFEF718B66147A7F4EEAED6E3B385025BADEA(a)wpf1malprd4.globeop.com> Content-Type: text/plain; charset="utf-8" I'm deploying Kamailio in an enterprise environment and I'm looking for a better way to handle the mapping of external DIDs to SIP usernames, number re-writing etc. I'm currently using ENUM in my test setup but I'd prefer to go with SQL storage and have that data written out to a faster database for lookups by Kamailio. I have a mix of users that register directly to the proxy and also several PBXs that I need to route blocks of numbers to through Cisco routers. For location A I would need to route any four digit number that starts with a 3 (3759), EXCEPT ones specifically defined in my database since they might be mapped directly to a user. Also for any outbound call (9 prefix) I need to check the DB to see if it's local. If I move to DB storage how do I handle things like parallel/sequential forking? ENUM makes it very easy - for example: 1.0.0.0.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:8700@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191986701####@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191996700####@172.20.152.132!" . If I have a SIP user "bob(a)domain.com" a simple location lookup works for that, but I also need to be able to map a e.164 did to him, and a short form with just the last 4 digits for my internal dialing. I would need a database schema that does the equivalent to this: 0.1.5.3.3.9.2.4.1.9.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . 0.1.5.3 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . Any suggestions? Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Serusers] Questions about strip() and altering ANI/From: numbers
by Frank Durda IV 13 Mar '09

13 Mar '09

When using strip() on a E.164 format number (eg +17895551212), does strip(2) delete the +1, or is it going to delete "17" in the shown example? I seem to be getting the latter behavior. If the latter is what is intended, how does one get rid of the plus? Second, is there something equivalent to strip() for editing the ANI/From: number? As I am doing PSTN gateway conversions, I have to strip the +1 from a E.164 ANI/From: (if sent) at some point so that the destination PSTN switch can correctly do the Calling Name database lookup. The North American Calling Name databases will only match if exactly ten digits are provided, so the ANI/From: has to be scrubbed down to ten digits if the country code in the ANI/From: begins with +1. Thanks!

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[Kamailio-Users] lcr-module, ping and load_gws_from_grp (kamailio 1.5)
by Stefan Prelle 12 Mar '09

12 Mar '09

Hi, I just noticed something weird when working with the lcr module. My setup has 5 gateways - 4 in group 1, one in group 2. The lcr-table has an entry for each group, both reacting on the same prefix. I am using load_gws_from_grp("1") to get a gateway for a target. This usually works fine. If now all 4 gateways in group 1 ar disabled because the alive check failed, load_gws_from_grp("1") returns the gateway from group 2 (or other groups where the prefix from the lcr-table matches. I would consider this a bug, since I explicitly requested gateways from a specific group. Regards, Stefan

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[Serusers] Problem with Record-Route not being added
by Frank Durda IV 12 Mar '09

12 Mar '09

I have recently noticed that despite having a $record_route_nated = true; in the route[PSTN_FORWARD] of ser.cfg, the INVITES and such sent on do not have a Record-Route header. In our NATed situation, probably the only thing this would cause problems with would be when the called party hangs up first, SER and the calling party never find out about it, since the calling party IP is not reachable from the PSTN switch NATed address space, so the single BYE message sent by the PSTN switch probably whizzes by to an unreachable destination address. So the calling party remains off-hook and just gets dead air. When the calling party finally disconnects, the SER passes the caller-initiated BYE to the PSTN switch which returns a 481 because it thinks the call was over ages earlier. Or at least this is what appears to be happening. Here is the PSTN_FORWARD in use, not that different from the PSTN gateway example ser.cfg: route[PSTN_FORWARD] { # Set where packets should be sent. If you don't do this here, # OPTIONS and other messages will be transmitted back to the # outside interface and processed all over again, and you eventually # get a Too Many Hops error on every OPTIONS message. # Code to make this point in multiple directions would be needed # to implement two-way origination. rewritehost("10.x.x.x"); #Send to SIP-TDM switch if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } else if (method == "INVITE") { # Force record routing if either party is behind NAT $record_route_nated = true; # log(1, "NAT Mangle Force RTP Proxy\n"); force_rtp_proxy("e"); } t_on_reply("PSTN_REPLY"); # if this is called from the failure route we need to open a new branch if (isflagset(FLAG_FAILUREROUTE)) { append_branch(); } # if this is an initial INVITE (without a To-tag) we might try another # (forwarding or voicemail) target after receiving an error if (method=="INVITE" && @to.tag=="") { t_on_failure("FAILURE_ROUTE"); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } drop; } The resulting INVITE that is being sent on to the PSTN switch looks like: INVITE sip:+1xxxxxxxxxx@10.x.x.x:5060;npdi=yes SIP/2.0 Via: SIP/2.0/UDP 10.x.x.x;branch=z9hG4bK6804.1cb4cb23.0 Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bK1kpu0b20e0fgogk9c6d0.1 f: <sip:+1xxxxxxxxxx;jip=xxxxxx@10.x.x.x:5060;user=phone>;tag=d0a-13c4-83e003-64bd22fa-83e003 t: <sip:+1xxxxxxxxxx@10.x.x.x:5060;user=phone> i: a249e890d8184d0a13c483e0033230d1ff3b57654c1d4018-0143-8166 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:+1xxxxxxxxxx;jip=xxxxxx@10.x.x.x;user=phone> Max-Forwards: 16 k: 100rel Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK m: <sip:10.x.x.x:5060;transport=udp> c: application/SDP l: 231 v=0 o=- 15109843 15109843 IN IP4 10.x.x.x s=- c=IN IP4 10.x.x.x t=0 0 m=audio 44464 RTP/AVP 0 101 b=AS:80 a=ptime:20 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=nortpproxy:yes So, any ideas as to why the "$record_route_nated = true" doesn't seem to be having any effect, eg no Record-Route shows up in outbound messages? I know SER is passing through the above code because I can uncomment the "NAT Mangle" comment and see it in the logs. Is there perhaps some module that must be present that we may have eliminated that makes Record-Route get emitted? Thanks again!

2 2

[Kamailio-Users] DID assignment & routing
by Robert McGilvray 12 Mar '09

12 Mar '09

I'm deploying Kamailio in an enterprise environment and I'm looking for a better way to handle the mapping of external DIDs to SIP usernames, number re-writing etc. I'm currently using ENUM in my test setup but I'd prefer to go with SQL storage and have that data written out to a faster database for lookups by Kamailio. I have a mix of users that register directly to the proxy and also several PBXs that I need to route blocks of numbers to through Cisco routers. For location A I would need to route any four digit number that starts with a 3 (3759), EXCEPT ones specifically defined in my database since they might be mapped directly to a user. Also for any outbound call (9 prefix) I need to check the DB to see if it's local. If I move to DB storage how do I handle things like parallel/sequential forking? ENUM makes it very easy - for example: 1.0.0.0.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:8700@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191986701####@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191996700####@172.20.152.132!" . If I have a SIP user "bob(a)domain.com" a simple location lookup works for that, but I also need to be able to map a e.164 did to him, and a short form with just the last 4 digits for my internal dialing. I would need a database schema that does the equivalent to this: 0.1.5.3.3.9.2.4.1.9.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . 0.1.5.3 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . Any suggestions? Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

2 1

[Kamailio-Users] SQLOps Module
by Andreas Heise 12 Mar '09

12 Mar '09

Hi, first of all thanks for the great sqlops module. Is "|sql_result_free(result)|" needed before I leave the script by "exit;" to avoid memory trouble or is it only needed to query again in the same request processing? Is there any way to check if the query was not successful (db unreachable, systax error in query)? I've seen a warning in the syslog, but I need to check it in the script. BTW in the doc the example for 1.4.2. seems to be wrong? regards, Andreas

2 3

[Kamailio-Users] Does Kamailio can be used as ptt server?
by doolin wu 12 Mar '09

12 Mar '09

Hi, I'm trying to find a open source ptt (push to talk) server? Does Kamailio can act as this role? Or do you know any open source ptt server? Thanks! -- Steven S.W

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[Kamailio-Users] Siremis: login problems
by Luntras Loredana 12 Mar '09

12 Mar '09

Helo, I've installes Siiremis and I have the following error when I try to log in. "An exception occurred while executing this script: Error message: #256, No Database information found in the config file. Script name and line number of error: /var/www/siremis-0.9.1/openbiz/bin/Configuration.php:42" I have only change the db_name in Config.xml from openser_siremis to openser. (the only data base that I have is openser that I've installed with opensedbctl create). Can anybody help me, please? Thanks, Loredana

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