sr-users April 2005

sr-users@lists.kamailio.org

250 participants
697 discussions

by raid＠gmx.net

hi list please can someone explain to use ser with rpid where is the best place to insert "append_rpid_hf("<", ">;party=calling;id-type=subscriber;screen=yes;privacy=off");" thanks regards raid -- +++ NEU: GMX DSL_Flatrate! Schon ab 14,99 EUR/Monat! +++ GMX Garantie: Surfen ohne Tempo-Limit! http://www.gmx.net/de/go/dsl

19 years, 6 months

[Serusers] PSTN => SER => PSTN

by Mik Cheez

I'm having an issue where my OUT PSTN is not getting a signal that a call has ended. My setup is as follows: IN OUT PSTN ==> SER ==> PSTN If I place a call and the person hangs up before the callee answers, the callee's phone just keeps ringing. I see a CANCEL event on the SER box, but nothing on the OUT PSTN. Here's a sample of my ser.cfg script: ------------------------------------ debug=3 fork=yes log_stderror=yes sip_warning=no check_via=no dns=no rev_dns=no port=5060 #children=4 fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/exec.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/xlog.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method!="REGISTER") { record_route(); }; xlog("L_INFO", "#INFO# %rm\n"); if (loose_route()) { t_relay(); break; }; setflag(1); if (method=="CANCEL"){ sl_send_reply("487", "Terminated"); break; }; if (method=="INVITE") { if (uri=~"sip:[0-9]+@.*") { xlog("L_INFO", "Original URI: %ru \n"); exec_dset("insert script here"); xlog("L_INFO", "Relay URI = %ru\n"); if (!t_relay()) { sl_reply_error(); break; }; }; }; }

19 years, 6 months

[Serusers] gw based on username

by Iqbal

19 years, 6 months

[Serusers] gw based of username

by Iqbal Gandham

Hi Currently I am using lcr to choose my gateway based upon prefix, now I wanted to choose a gateway based on username. I was thinking that if a user made a call to say 001xyz I could detect its username, see if in usr_prefences (using avp) it had a variable set eg:add_prefix, if so add a prefix of 2 to the number. Then in lcr add a false prefix of 2001 to the table with the new gateway, and at the gateway remove the prefix, before making the call, this way I dont have to ask users to dial any special prefixes. Any better ideas -- Regards Iqbal Gandham, CEO IPClouds Ltd (http://www.ipclouds.co.uk) Baylis House ,Stoke Poges Lane Slough,Berkshire SL1 3PB V: iqbal(a)sip.ipclouds.co.uk ( call me free !!) M: +44 7866318220 F: +44 (0) 870 7474784 W: http://www.ipclouds.co.uk ------------------------- b5 18 b5 07 1c cd 15 41 ba 23 ad 94 38 30 b8 df -------------------------

19 years, 6 months

[Serusers] AS 5350 + Ser Problem

by Razvan Nemesiu

Hi all, I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming pots dial-peer from PSTN (let's say that it matches 333XXX numbers).Then I have two outgoing voip dial-peers (both dial-peers match 333XXX numbers and are the outgoing routes for these numbers).One of these two outgoing dial-peers is set with a higher preference than the other in order to be the first choice for sending the calls.One of the dial-peers has session target one of the SER and the other has the other SER as session target.So, the main idea is that I want to do fallback between these two dial-peers: if the connection with the first SER (that is with higher preference) is down or there are network problems I want my CISCO to choose the other route to my second SER (and I want it to do that after a specified time).The question is how cand I do that ? And how can I set this timer ?(let's say 10 seconds -> after 10 seconds to choose the other route).What commands do I have to set on the dial-peers and what commands need to be set in the global configuration (timers,etc...)? Thank you very much and I appreciate any help.

19 years, 6 months

[Serusers] Help with authorization

by Steve Blair

I have a working 0.9.1 config to which I would like add server side features such as call forward all (cfwdall). I have a pretty good idea how to handle cfwdall using avp_ops however I'm stuck on the authentication. If a subscriber has local calling permissions (acl=local) and cfwdall their phone to a long distance number I need to respond with an informative response. In the INVITE processing in my config I have statements such as: if (is_user_in("credentials", "ld")) { setflag(11); }; These checks fail with the following errors: Apr 21 18:31:53 ser[498]: [SER]: AVP: Checking From gateway caller Apr 21 18:31:53 ser[498]: check_username(): No authorized credentials found (error in scripts) Apr 21 18:31:53 ser[498]: check_username(): Call {www,proxy}_authorize before calling check_* function ! Apr 21 18:31:53 ser[498]: [SER]: Flag for UMVM redirect successful. Apr 21 18:31:53 ser[498]: [SER]: AVP: Checking credentials Apr 21 18:31:53 ser[498]: is_user_in(): No authorized credentials found (error in scripts) I thought adding proxy_authorize("", "subscriber")), check_to and check_from calls prior to the is_user_in check would address these errors but that hasn't worked either. If I want to set a flag if the caller is an authorized subscriber, the callee is an authorized subscriber and then use "is_user_in" to determine if the called party has a particular credential what am I missing? Thanks,Steve

19 years, 6 months

[Serusers] serweb path

by Llanos Serna García-Conde

19 years, 6 months

[Serusers] SER 0.8.12 malformed outgoing packets with some UA

by Arek Bekiersz

Dear List, Some new UA device we are testing is causing strange behaviour of SER. When INVITE is forwarded by our SIP proxy (using SER 0.8.12 cvs + Radius backend), it contains some garbage characters and IP packet lenght is disrupted. Packet is cut on exact lenght of 1478 bytes (some MTU limit of 1500 bytes here???) See little debug below (I can send full debug on priv address). Notice the additional "."'s before INVITE and garbage in address line (U xx -> zz:5060 414@0:1480): I double checked SER log in debug mode, there are no differences between this abnormal situation and normal INVITE proxying, with other UA devices. Initial buffers are set to high values and nothing can go wrong.. Is this some bug in SER? Note: adresses were changed to xxx,yyy and so on: U xx -> yy:5060 SIP/2.0 100 trying -- your call is important to us.. Via: SIP/2.0/UDP yy:5060;branch=z9hG4bKDB20B3494FE0C715E1DA2AB3390D.. From: <sip:vvv@xx>;tag=6817B284750F6B75621571A17923.. To: <sip:www@xx>.. Call-ID: E5A4F1BBFB9D021B58C6B6440CB3@yy.. CSeq: 18 INVITE.. Server: Sip EXpress router (0.8.12-1rc6 (i386/freebsd)).. Content-Length: 0.. Warning: [...omited...] # U xx -> zz:5060 414@0:1480 <-- NOTICE garbage here ........INVITE sip:www@xxx SIP/2.0.. <-- NOTICE "dots" here Record-Route: <sip:www@xxx;ftag=6817B284750F6B75621571A17923;lr=on>.. Via: SIP/2.0/UDP xxx;branch=z9hG4bK8bef.e7a84e21.0.. Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bKDB20B3494FE0C715E1DA2AB3390D.. From: <sip:vvv@xxx>;tag=6817B284750F6B75621571A17923.. To: <sip:www@xxx>.. Call-ID: E5A4F1BBFB9D021B58C6B6440CB3@yyy.. CSeq: 18 INVITE.. Contact: <sip:vvv@yyy;uniq=BE4F86E6E5C314AE11701C942248>.. Authorization: [...omited...] Max-Forwards: 69.. User-Agent: [...omited...] Supported: 100rel, replaces.. Allow-Events: telephone-event.. Allow-Events: refer.. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER.. Content-Type: application/sdp.. Accept: application/sdp.. Accept-Encoding: identity.. Content-Length: 418.. .. v=0.. o=user 20040204 20040204 IN IP4 yyy.. s=call.. c=IN IP4 yyy.. t=1114074899 1114078499.. m=audio 7078 RTP/AVP 8 0 2 102 100 99 18 101 13.. a=rtpmap:8 PCMA/8000.. a=rtpmap:0 PCMU/8000.. a=rtpmap:2 G726-32/8000.. a=rtpmap:102 G726-32/8000.. a=rtpmap:100 G726-40/8000.. a=rtpmap:99 G726-24/8000.. a=rtpmap:18 G729/8000.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-11.. a=rtpmap:13 CN/8000.. a=rt <--- NOTICE packet is cut here at 1478 bytes ######################################## U zzz:5060 -> xxx:5060 SIP/2.0 100 Trying.. Via: SIP/2.0/UDP xxx;branch=z9hG4bK8bef.e7a84e21.0.. Via: SIP/2.0/UDP yy:5060;branch=z9hG4bKDB20B3494FE0C715E1DA2AB3390D.. From: <sip:vvv@xxx>;tag=6817B284750F6B75621571A17923.. To: <sip:www@xxx>;tag=as7a954f71.. Call-ID: E5A4F1BBFB9D021B58C6B6440CB3@yyy.. CSeq: 18 INVITE.. User-Agent: [...omited...].. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:www@zzz>.. Content-Length: 0.. .. Here is tcpdump of this situation (note the packet lenght error): --------------------------------------------------------------------------- Packet 5 Timestamp: 01:00:07.969419 (0.068520) Source Ethernet Address: 00:00:5E:00:01:01 (<unknown>) Destination Ethernet Address: 00:E0:1E:DF:AE:82 (<unknown>) Encapsulated Protocol: IP IP Header Version: 4 Header Length: 20 bytes Service Type: 0x10 Datagram Length: 614 bytes Identification: 0x45FE Flags: MF=off, DF=off Fragment Offset: 0 TTL: 64 Encapsulated Protocol: UDP Header Checksum: 0xDF42 Source IP Address: xxx Destination IP Address: yyy UDP Header Source Port: 5060 (<unknown>) Destination Port: 5060 (<unknown>) Datagram Length: 594 bytes (Header=8, Data=586) Checksum: 0xB306 UDP Data SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bK003971715 B46D0300099FFA02356. From: <sip:vvv@xxx>;tag=3085BFD5D9E8D8717CD 04F54A2A5. To: <sip:www@xxx>. Call-ID: 37C1AE7089D5A7FA8302B724D791@yyy. CSeq: 13 INVITE. Server: Sip EXpress router (0.8.12-1rc6 (i386/freebsd)). Content-Length: 0. Warning: [...omited...]. . --------------------------------------------------------------------------- Packet 6 Timestamp: 01:00:07.969683 (0.000264) Source Ethernet Address: 00:00:5E:00:01:01 (<unknown>) Destination Ethernet Address: 00:0C:6E:A5:38:55 (<unknown>) Encapsulated Protocol: IP IP Header Version: 4 Header Length: 20 bytes Service Type: 0x10 Datagram Length: 1500 bytes Identification: 0xCBCD Flags: MF=on, DF=off Fragment Offset: 0 TTL: 64 Encapsulated Protocol: UDP Header Checksum: 0x3A55 Source IP Address: xxx Destination IP Address: zzz <*** Packet length corrupt ***> --------------------------------------------------------------------------- --------------------------------------------------------------------------- Packet 7 Timestamp: 01:00:07.971393 (0.001710) Source Ethernet Address: 00:0C:6E:A5:38:55 (<unknown>) Destination Ethernet Address: 00:00:5E:00:01:01 (<unknown>) Encapsulated Protocol: IP IP Header Version: 4 Header Length: 20 bytes Service Type: 0x10 Datagram Length: 538 bytes Identification: 0x00DE Flags: MF=off, DF=on Fragment Offset: 0 TTL: 64 Encapsulated Protocol: UDP Header Checksum: 0xE906 Source IP Address: zzz Destination IP Address: xxx UDP Header Source Port: 5060 (<unknown>) Destination Port: 5060 (<unknown>) Datagram Length: 518 bytes (Header=8, Data=510) Checksum: 0xF96E UDP Data SIP/2.0 100 Trying. Via: SIP/2.0/UDP xxx;branch=z9hG4bK487e.541ff4c2 .0. Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bK003971715 B46D0300099FFA02356. From: <sip:vvv@xxx>;tag=3085BFD5D9E8D8717CD 04F54A2A5. To: <sip:www@xxx>;tag=as2f2ca5dd. Call-ID: 37C1AE7089D5A7FA8302B724D791@yyy CSeq: 13 INVITE. User-Agent: Perceval Voicemail Server Idun. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:www@zzz>. Content-Length: 0. . Thanks for help... Regards, Arek Bekiersz arek(a)perceval.net ------------------------------------------------------------------------ Perceval Technologies sa/nv Rue Tenbosch, 9 B-1000 Brussels Tel: +32-2-6409194 (ext.213) Fax: +32-2-6403154 URL: http://www.perceval.net/ E-mail for general informations: info(a)perceval.net E-mail for technical informations: helpdesk(a)perceval.net ------------------------------------------------------------------------ This e-mail message contains legally PRIVILEGED and CONFIDENTIAL information intended for the use of the addressee only. If you are not the intended recipient of this message, please notify the undersigned by telephone or e-mail reply and destroy this message and any attachments. Any views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of Perceval. The integrity and security of this message cannot be guaranteed and it may be subject to data corruption, interception, unauthorised amendment, viruses and unforesee

19 years, 6 months

[Serusers] Help Using SER FIFO "ser_fifo"

by Leo

Dear serusers, I have spent the past several nights trying to figure out how to use the ser_fifo to send commands and get info back from SER using PHP. My understanding of using the fifo in SER is to send it :fifo_command:result_filename. In the example below I am sending the "ul_dump" command to the fifo and telling to return the result to a file called "delthis.txt". The file "delthis.txt" never gets created and it doesn't look like this fifo thing is working at all. The examples in the admin guide are not helping me. Can anyone one help? The test PHP code is below: <html> <head> <title>Browser SERCTL </title> </head> <body> <?php function wrfifo() { $fifo_handle = fopen("/tmp/ser_fifo" , "w"); if (!$fifo_handle) { echo "cannot open ser_fifo<br>"; return 0; } if (fwrite($fifo_handle, ":ul_dump:/tmp/delthis.txt")==-1) { echo "cannot write to fifo"; fclose($fifo_handle); return 0; } fclose($fifo_handle); system("cat /tmp/delthis.txt"); return 1; } echo "begin<br>"; $success = wrfifo(); if ($success) { echo "success<br>"; } else { echo "failure<br>"; } ?> Leo Papadopoulos leo(a)ltcjp.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.18 - Release Date: 4/19/2005

19 years, 6 months

[Serusers] Service Creation Options

by Aisling O'Driscoll

Hello, I am looking to develop services based on the SER platform which can be configured via a secure web interface e.g. sequential ringing, subscriber based call routing etc. It seems that the service languages available are: 1. CPL 2. SIP CGI 3. SIP Servlets Basically Im a little bit confused as to which one I should target. I believe CPL is for use on end user terminals so Im ruling that out. I was thinking of sip servlets...Any opinions on this?...Will this be able to manipulate SER easily and also is SerWeb in any way linked to this?....Would it be possible to use a combination of servlets and CPL?? Any information would be appreciated, Many Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

19 years, 6 months

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sr-users April 2005