sr-users June 2009

sr-users@lists.kamailio.org

80 participants
112 discussions

[Kamailio-Users] Media information in stateful mode
by karhu 16 Jun '09

16 Jun '09

Hello, I'm back with Kamailio ;) I would like to know if there is a issue to store media informations, like codecs, Ip, ports of a the caller and the callee.. I've been looking to TM module, but it doesn't seem to be possible to manipulate media parameters !? Thanks in advance, Karhu -- View this message in context: http://www.nabble.com/Media-information-in-stateful-mode-tp24050009p2405000… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Firewall and auth
by Andreas Granig 15 Jun '09

15 Jun '09

Hi all, I just tried a setup like [UA] --> [pub][Firewall][priv] --> [priv][Kam] where the Firewall maps the public IP reachable by UAs to a private IP where Kamailio is listening. If I run sipsak on the Kamailio-machine, I can register fine, but as soon as the request goes via Firewall, authentication stops working. So how does the IP of Kamailio actually influence authentication? Do I have to set something special on Kamailio to make this work? Here's the Register after a 401 and the resulting 401 again, and it looks pretty well to me (1.2.3.4 is the public Firewall IP, which is configured as outbound proxy on the UA, 172.17.10.50 is the private Kamailio-IP and is also used as domain for user sipwise1, which is trying to register). Trace is taken on client-side, but looks the same on the Kamailio server (NAT seems to be handled fine): U 192.168.123.150:50600 -> 1.2.3.4:5060 REGISTER sip:1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.123.150:50600;rport;branch=z9hG4bK906580090. From: <sip:sipwise1@172.17.10.50>;tag=1631756043. To: <sip:sipwise1@172.17.10.50>. Call-ID: 1235449552. CSeq: 4 REGISTER. Contact: <sip:sipwise1@192.168.123.150:50600;line=e779ddd40d3251b>. Authorization: Digest username="sipwise1", realm="172.17.10.50", nonce="4a06e2820000000a80c173db2d166fedb7d8d1e933c97855", uri="sip:1.2.3.4", response="de645a701a7c507c47a5278923bce54b", algorithm=MD5. Max-Forwards: 70. User-Agent: Linphone/2.1.1 (eXosip2/3.1.0). Expires: 900. Content-Length: 0. U 1.2.3.4:5060 -> 192.168.123.150:50600 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.123.150:50600;rport=50600;branch=z9hG4bK906580090;received=213.47.175.165. From: <sip:sipwise1@172.17.10.50>;tag=1631756043. To: <sip:sipwise1@172.17.10.50>;tag=a49efde55ae28efd11dc5969af09c5db.b607. Call-ID: 1235449552. CSeq: 4 REGISTER. WWW-Authenticate: Digest realm="172.17.10.50", nonce="4a06e2820000000b2bd307dd3e71c80e3d6549ccc2b28269". Server: Sipwise registrar. Content-Length: 0. So the only thing referring to the public Firewall IP is in the R-Uri of the registration and in the Authorization-uri-token. Is this token also used to calculate the auth hashes somehow? Username looks fine in the Authorization header, and so does Realm. Any ideas? Andreas

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[Kamailio-Users] FW: /etc/init.d/kamailio start can't start
by vinod 15 Jun '09

15 Jun '09

Dear Saúl Ibarra I don't modify my configuration file, but I using "'kamailio'" to start ,then i got the msg : Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 140.125.35.240 [140.125.35.240]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 140.125.35.240 [140.125.35.240]:5060 Aliases: tcp: vino.local:5060 tcp: localhost:5060 udp: vino.local:5060 udp: localhost:5060 is it represents the openser success ? cheers, vinod -----Original Message----- From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Saul Ibarra Sent: Tuesday, June 09, 2009 3:21 PM Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] /etc/init.d/kamailio start can't start Try this options in your configuration file: debug=6 fork=no log_stderror=yes And start kamailio just by executing 'kamailio', without the init script, so you'll be able to see what happens. -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/ _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] log_facility usage with accounting module
by Örn Arnarson 15 Jun '09

15 Jun '09

Hi, I am trying to get all billing information from the acc module logged to a file as opposed to a database. It seems to me that the way to do that is via syslog (although I'd much prefer just logging directly to a file). So, to do this, i added a LOCAL6 facility in /etc/syslog.conf, and added the following line to kamailio.cfg: modparam("acc", "log_facility", "LOG_LOCAL6") The general log_facility of kamailio is LOG_LOCAL7: log_facility=LOG_LOCAL7 These are the only two mentions of LOG_LOCAL6 and LOG_LOCAL7 in the whole configuration file. However, I get _all_ the information in both files (that is, if I have debug set to a specific level, my accounting log also gets flooded with these debug messages). Why is this? The config line specifies LOG_LOCAL6 to be a log for the accounting module... Furthermore, I want all INVITES to be logged, regardless of an ACK or CANCEL, but only invites where an ACC is received are logged. If I enable log cancel, I see the CANCEL messages, but not the INVITES, which is rather nonsensical. Any help would be much appreciated. Best regards, Örn

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[Kamailio-Users] Dynamic mapping of sip URI
by rachid 15 Jun '09

15 Jun '09

Hello, i use the database table purplemap of kamailio to mapping sip addresses with jabber uri, Now i want to use a dynamic mapping of uri sip such alice(a)jabber.com becomes sip:alice-at-jabber@domain_sip someone have an idea to do that? -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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[Kamailio-Users] sharing audio between a SIP and a MSN client
by rachid 15 Jun '09

15 Jun '09

Hello, I installed kamailio and i want to have sharing audio between a sip client and MSN client , but i don't know which module i will to configure someone can tell me what i do please ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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[Kamailio-Users] bidirectionnal chat and presence sharing between a SIP and a jabber user
by rachid 15 Jun '09

15 Jun '09

Hello all, someone can help me to configure kamailio to i was able to have a bidirectionnal chat and presence sharing between a SIP and a jabber user ? thank you

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[Kamailio-Users] Munin plugin for Kamailio
by Iñaki Baz Castillo 15 Jun '09

15 Jun '09

Hi all, I've created a Munin plugin to monitor the number of dialogs in Kamailio: http://dev.sipdoc.net/wiki/sip-stuff/KamailioMuninPlugin Hope it's useful. Suggestions are welcome. Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Presence and jabber
by Joao Gomes Pereira 15 Jun '09

15 Jun '09

Hello to all I would like to enable Presence support and Jabber integration for my Kamailio users, but I'm a bit confused. I already installed Jabberd2. So, why do I need to enable presence support in Kamailio? Presence in Kamailio is only for VoIP clients that doesnt support standard Jabber Instant Messaging? Thanks Joao Pereira

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[Serusers] Is there a way to force the added Via: to contain a specific IP addr != send socket
by Frank Durda IV 15 Jun '09

15 Jun '09

I have a configuration where SER sits immediately adjacent to a NAT translation between 10.x.x.x on the SER side and public Internet on the other side of the NAT. This a PSTN gateway that has only handled inbound SIP traffic, but now I am trying to send outbound and have run into issues. I am able to fix Contact: and SDP payload IP addresses and such with various force commands, but I have not found a way to alter Via:, and that is causing problems. To get the INVITE to forward out the right interface, we have to do a force_send_socket of say 10.2.4.6 which does get the packet out on the right interface headed toward the NAT router, but that same value also gets put in the Via: header and so none of the public IP address destinations (eg 208.22.33.44) can send SIP messages back to our public IP address gateway (205.66.77.88). So the INVITE that reaches 208.22.33.44 has packet source of 205.66.77.88), but the Via: is 10.2.4.6 and 208.22.33.44 replies to the Via: address, so we don't get any of the responses. So, is there a way to force the Via: to put out the needed value so that things will come out right?

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sr-users June 2009