sr-users October 2003

sr-users@lists.kamailio.org

105 participants
230 discussions

by Iizuka, Hirotaka, VF-JP

Hi, Does SER hold registered SIP clients' addresses? I use DHCP server to address IPaddresses to SIP clients in my small SIP network. The diagram of the topology is shown below; DHCP -----------------+ | Client A ------------+ | | | Hub | | Client B ------------+ | | SER --------------------+ In this case, I don't have a DNS server yet. I wanted to know if DNS is necessary here. Thnak you. Hiro

21 years, 1 month

[Serusers] dbtext module

by Annie Sasidar

Hi, I have some questions about the dbtext module- I loaded the module and have written a text file table with user information in the format specified in the documentation. How do i check whether the table information is being used? the client is Windows messenger 4.7. I get a 404 error when i try to invite a user to start a conversation.Also, the documentation mentions about an API specified by the SER DB interface. Where do i get this API? I cannot administer with serctl now since it does not keep a record of all the users registered, it keeps record only of current user. It would be great if someone who has played with this module give me some tips. Thanks, Annie

21 years, 1 month

[Serusers] About ser.cfg

by giAndrea

Hi to all I would like to thank all the people that in this months help me to using ser. I have a question or better a doubt . From june to now I've tried to configure my ser writing ser.cfg file. I've read the admin_guide manual and i learn how to use command and modules, but i don't understend how to write a correct ser.cfg. I try to explain better. A file ser.cfg is executed for received request write to route some request and every one are processed, i can receive PSTN request, REGISTER request or an INVITE. But, what's the order on ser.cfg to process this request? PSTN is more important than SIP request? Outbound request before Inbound? My actual situation is: my pc <--> NAT <--> Sip Proxy <--> NAT <--> Other User (PSTN or SIP) od my pc <--> NAT <--> Sip Proxy <--> Other User (PSTN or SIP) (At the moment i haven't implemented PSTN support on ser.cfg) I actually use this scheme (it's simil;ar than default ser.cfg): - initial sanity checks - record route and loose route processing - process some enc_prefix request - process some private request (sdp_mangler for INVITE request and encode_contact for the other) - if user are from myself (uri == myself), register if is a REGISTER request or lookup and after t_relay - else t_relay outbound request without considering the technical problems, is this the correct semantic to use? Where have I to process PSTN and VoiceMail request? Thanks, Andrea ------------------------------------------------------- Andrea Girardi mobile +39 347 9624051 sip address: giandrea(a)sip.quellidelpirana.it http://www.giandrea.com

21 years, 1 month

[Serusers] SER Contact rewrite for Softswitch

by Harry Behrens

Hello, we have the following setup in our "Unified Messaging lab": SUAs <-> LAN <-> SER <-> Softswitch <-> PSTN The SER is provisioned as a SIP Server (essentially a SUA) from the Softswich point of view. We have provisioned a number block of 20 numbers (from a 30-channel ISDN PRI) for direct inward dialling (DID) to the SER. E.g. 12345620-12345639 are routed to the SER. On the SER we have a number of users <user>@dai-labor.de, where the first 20 users have an alias of 123456[23][0-9] Each user also has his number added to his permanent contacts to enable (on the cheap) a "call by PSTN if not online" poliy. The aim obviously being to have a 1-1 mapping between URI and allocated telephone number (we are still in the proof-of-concept phase and will move to DNS and ENUM based architecture at a later stage). Scenario 1: OK PSTN user Up calls 12345621 - which is mapped to harry(a)dai-labor.de. If harry(a)dai-labor.de is registered, his SIP UA rings and the call is established. Scenario 2: problem harry(a)dai-labor.de wants to use his UA to place a call through the Softswitch. The Softswitch "authenticates" (if you can call this authentication ;-( by looking at the Contact header and comparing it to its database of numbers assigned to provisioned SIP Servers. I therefore need to be able to specify a rewrite of the Contact header to replace harry@<Contact IP:port> with 12345621@<Contact IP:port> when the callee is of the form <number>@<whatever> I am aware that this would be strictly speaking not proxy but rather B2BUA behaviour, but would still be interested to know whether anyone has a solution to this problem or whether we need to hack one ourselves. Many thanks in advance, Harry Dr. Harry Behrens Projektleitung BIB3R DAI Labor - Technische Universität Berlin Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin Fon: +49 30 314 23383 Fax: +49 30 314 21799 Email: harry.behrens(a)dai-labor.de http://www.dai-labor.de

21 years, 1 month

[Serusers] Upgrade from 0.8.10 to 0.8.11

by Lennart Helleberg

Do I have to uninstall version 0.8.10 to upgrade to 0.8.11? I'm using the mysql-module. If I tried to upgrade with an rpm package, but I got dependence problem between ser and ser-mysql module. Anybody have some best practise info?

21 years, 1 month

[Serusers] uri==myself check and record_route()

by Maxim Sobolev

Folks, I am correct that it is impossible to use uri==myself check along with record_route()? That is, consider the following situation: UA1->SER (with record_route())->UA2 UA1 sends INVITE to SER: INVITE sip:ua2@ser.ip Contact: <sip:ua1@ua1.ip> After processing SER resends it to UA2: INVITE sip:ua2@ua2.ip Record-Route: <sip:ua2@ser.ip;lr> Contact: <sip:ua1@ua1.ip> Then, if UA2 wishes to terminate session it sends the following BYE to the SER: BYE sip:ua1@ua1.ip Route: <sip:ua2@ser.ip;lr> Contact: <sip:ua2@ua2.ip> Therefore if there any if (!(uri==myself)) { sl_send_reply("500", "example serves only my domain"); break; }; checks in the SER, it will reply with 500 to UA2, not allowing transaction to go through. -Maxim

21 years, 1 month

[Serusers] charset

by Teppo Lunnas

Hi, I tried to change charset to ISO 8859-10 in both send_im.php and config.php -directories, but it didn't help. The scandinavian chars couldn't be sent. Teppo

21 years, 1 month

[Serusers] charset

by Teppo Lunnas

Hi, I am trying to send IM with serweb to Messenger 4.7, but I can't use scandinavian characters. In send_im.php the default set was UTF-8 and I changed it to ISO-8859-1 or ISO-8859-10 but it didn't help. So what should I do to get the serweb accept scandinavian characters?

21 years, 1 month

[Serusers] Click to dial

by Teppo Lunnas

Hello, I have successfully installed SER 0.8.11 and serweb. There is still one feature in serweb that doesn't work in my installation. That new feature is the possibility to start a SIP-session from the personal phonebook. I have two PCs in the same network and both have Messenger 4.7 clients. When their SIP-addresses in the serwebs phonebook are online and I click the one I want to call to there comes a message "488 Not Acceptable Here". What goes wrong?

21 years, 1 month

[Serusers] FIN,ACK right after a successful registration

by ahmet taskeser

Hi Serusers, I am suffering from a weird problem in SER. Maybe you would help me with that. I have SER 0.8.11(TCP Support) on Linux Redhat v9. I am running our SIP user agent emulator against SER in my configuration. SER proxy is sending "FIN, ACK" to the user agent right after the REGISTER (200 OK). Although there is one established TCP connection and a successful registration, SER is terminating the TCP connection. Any help would appreciate Many thanks in advance Ahmet _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus

21 years, 1 month

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sr-users October 2003