sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[Serusers] Issues with SER for TCP load testing
by Stephen McVarnock 30 Nov '05

30 Nov '05

Hi folks, I am new to the mailing lists - unsure as to which list is the most appropriate I have mailed to both - I apologise for this in advance! I am using SIPP and SER(v 0.9.4) to load test an application - SIPP is sending calls via UDP to SER at 15 cps and with a duration of 180 seconds. SER is then forwarding on the calls to the application using TCP and receiving the replies back over TCP. However I am seeing quite a few errors in SER after a while. I'd like to determine if the issues I am seeing are with SER or with the application itself so I'd appreciate some help with the following errors. Can someone tell me, for each of the error traces below, what is the root cause of it and (even better) how I can resolve it for SER - if it is indeed an issue with SER and not with the application. 1) 123(14715) ERROR: tcpconn_new: mem. allocation failure 123(14715) ERROR: tcp_main_loop: tcpconn_new failed, closing socket 2) 54(14646) ERROR: tsend_stream: failed to send: (32) Broken pipe 54(14646) ERROR: tcp_send: failed to send 3) 65(14785) ERROR: tcp_blocking_connect: SO_ERROR (146) Connection refused 65(14785) ERROR: tcpconn_connect: tcp_blocking_connect failed 65(14785) ERROR: tcp_send: connect failed 65(14785) msg_send: ERROR: tcp_send failed 4) 63(14783) ERROR: tsend_stream: send timeout (10000) 63(14783) ERROR: tcp_send: failed to send 5) There was also another error on the TCP select loop - select failed: Too many connections (or something similar). I have been playing around with the number of descriptors set using limit but without success. Like I said, my main concern is to eliminate SER where possible as the cause of these issues before focusing attention on the application. Thanks in advance, Steve.

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[Serusers] Creating CDRs from SERs accouning records
by Adrian Georgescu 30 Nov '05

30 Nov '05

Frank, you can download CDRTool from: http://cdrtool.ag-projects.com/ CDRTool it is not publicly supported, we provide support only for our customers but you may give it a try. Adrian >>>>>>>>>>> Hi all i'm looking for a free/opensource application that generates CDRs (not XDRs, must not be rated) from the accounting data/events generated by SER (i think this is usually called normalization). I think this would have to be an application that is able to match INVITEs and BYEs and form CDRs with call start timestamp, call duration and so on. I had a look at the list of accounting software on the iptel.org website, but most of there applications are highly integrated accounting/billing solutions, far more than what i need (and also most of them seem to be commercial products). In the mailinglist archive i only found some references to CDRTool which seems also to be commercial. Obv. there must be a free version of CDRTool but neither i'm sure whether it implements exactly what i need so i could "grab it" for my own purposes nor could i find a place to download the free version). Does anyone know if there is such a kind of application available? Or any other idea how to deal with my requirements? Thanks a lot for your help Frank

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[Serusers] REGISTER message headers verification
by Aisling O'Driscoll 30 Nov '05

30 Nov '05

Hi, I was wondering if someone could please cast their eyes over the register sent from one of my phones behind a nat and the 200 OK sent back from SER. There has been an issue with no audio between a natted client and a public client. I looked at the invite and noticed the private address was being used in the Contact header field and the c field in the sdp so obviously the headers are not being rewritten properly....I then checked the register messages to see if the phone was registering correctly - The message sequence is below. I notice the private header is shown but that would probably be correct seeing fix_nated_register doesnt actually change the header, it simply adds fields, is that right? i.e. thats why it replaced fix_nated-contact to be more RFC compliant Anyway its the contact header fields sent back in the 200OK that I would like verified - Should the private address be listed as one of the options here? I have gone back to the onsip feature callfwd script using mediaproxy to eliminate any errors due to changes I may have made The ONLY difference between the scripts is that I have if(uri != myself) { route(4); route(1); break; }; instead of #if (!is_uri_host_local()) { # if (is_from_local() || allow_trusted()) { # route(4); # route(1); # } else { # sl_send_reply("403", "Forbidden"); # }; # break; # }; as otherwise the phones wouldn't register. Many thanks, Aisling. SER: 157.190.74.152 Phone Private Address: 172.16.3.13 NAT Box: 84.203.148.14 U 84.203.148.14:5060 -> 157.190.74.152:5060 REGISTER sip:157.190.74.152:5060 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.13;bran ch=z9hG4bK7543671c6ca1f07a..From: "Aisling 2092" <sip:2092@157.190.74.152:5 060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Contact: <si p:2092@172.16.3.13>..Call-ID: 78f22336c383663f@172.16.3.13..CSeq: 100 REGIS TER..Expires: 3456000..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK7543671c6ca1 f07a;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190.74.152: 5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Call-ID: 78 f22336c383663f@172.16.3.13..CSeq: 100 REGISTER..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 " Noisy feedback tells: pid=1742 req_src_ip=84.203.148.14 req_src_port=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1". ... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK754367 1c6ca1f07a;rport=5060;received=84.203.148.14..From: "Aisling 2092" <sip:209 2@157.190.74.152:5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5 060>;tag=4b358c93636b59b9f78eca99791ec991.cdd2..Call-ID: 78f22336c383663f@1 72.16.3.13..CSeq: 100 REGISTER..WWW-Authenticate: Digest realm="157.190.74. 152", nonce="438dfc0821fd70b65dbd9c0db3e5aa29fa1fcd82"..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.15 2:5060 "Noisy feedback tells: pid=1742 req_src_ip=84.203.148.14 req_src_po rt=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_ cnt==1".... # U 84.203.148.14:5060 -> 157.190.74.152:5060 REGISTER sip:157.190.74.152:5060 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.13;bran ch=z9hG4bK9e94983538147a54..From: "Aisling 2092" <sip:2092@157.190.74.152:5 060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Contact: <si p:2092@172.16.3.13>..Authorization: Digest username="2092", realm="157.190. 74.152", algorithm=MD5, uri="sip:157.190.74.152:5060", nonce="438dfc0821fd7 0b65dbd9c0db3e5aa29fa1fcd82", response="90d926c009567996b99e76fef560b9c6".. Call-ID: 78f22336c383663f@172.16.3.13..CSeq: 101 REGISTER..Expires: 3456000 ..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,AC K,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK9e9498353814 7a54;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190.74.152: 5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Call-ID: 78 f22336c383663f@172.16.3.13..CSeq: 101 REGISTER..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 " Noisy feedback tells: pid=1744 req_src_ip=84.203.148.14 req_src_port=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1". ... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK9e94983538147a54 ;rport=5060;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190. 74.152:5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>;tag=4 b358c93636b59b9f78eca99791ec991.3e0a..Call-ID: 78f22336c383663f(a)172.16.3.13 ..CSeq: 101 REGISTER..Contact: <sip:2092@157.190.74.151>;expires=3279337, < sip:2092@172.16.3.13>;expires=3447275;received="sip:84.203.148.14:50 60", <s ip:2092@84.203.148.14:5060>;expires=3456000..Server: Sip EXpress router (0. 9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 "Noi sy feedback tells: pid=1744 req_src_ip=84.203.148.14 req_src_port=5060 in_ uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1".... ##exit 91 received, 0 dropped localhost:~ # -------------------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.

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[Users] AVPOP and openser DB key
by Matteo Piazza 30 Nov '05

30 Nov '05

I use avpops module. I use the user_preferences table (openser DB) as avp_table: modparam("avpops","avp_table","usr_preferences") But I have a question: In the description of avp_table is not required a key, in the default configuration of user_preferences table there is a key composed by user-domain-attribute. But with this key I can't use some different value for the same attribute: for example "fwdbusy". So I have delete this key, and the program is ok. Someone knows why in the user_preferences the key is composed those three values -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

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Re: [Users] How to Configure a Redirect Server Using OpenSer
by Bogdan-Andrei Iancu 30 Nov '05

30 Nov '05

Hi, do you get any error? do you see the 3xx reply? are you sure the script execution gets there? regards, bogdan Ajay Srivastava wrote: >Hi bogdan, >Thanks for ur suggestion. I made changes in openser.cfg as per the example >which you have given and then restart openser, but still not getting >redirect messages, should I need to do something more for enabling this >option? Please help me and tell me.. >Regards, >ajay > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Monday, November 28, 2005 5:02 PM >To: Ajay Srivastava >Cc: users(a)openser.org; Aviraj Saha >Subject: Re: [Users] How to Configure a Redirect Server Using OpenSer > >Hi Ajay, > >there is a script example on the cvs: > >http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/examples/redirect.c >fg?rev=1.2&view=auto > >regards, >bogdan > >Ajay Srivastava wrote: > > > >>Hi >> >>We are trying to configure OpenSer as Redirect Server. Can anyone >>provide me with a configuration file that explains what options we >>have in OpenSer to configure it as redirect server. >> >> >> >>With Regards >> >>Ajay >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >>

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[Users] uac_replace_from errors
by Patrick Jordan-Smith 30 Nov '05

30 Nov '05

Hi there, I'm not totally sure if this is a problem with SER or not, but I have a interesting problem when using uac_replace_from together with LinkSys PAP2 ATAs (and possibly other linksys equipment, retesting tonight.) When you configure a name in the linksys ATA, you can put your name, ie, for myself I put Patrick Jordan-Smith. The linksys sends this like: From: Patrick Jordan-Smith <sip:728728@sip.qsi.net.nz> When I try to use uac_replace_from with this from address it chokes on it: ERROR: parse_to : unexpected char ["] in status 8: <<Beetlestone >> . ERROR:parse_from_header: bad from header ERROR:uac:restore_from_reply: failed to find/parse FROM hdr ERROR: parse_to : unexpected char ["] in status 8: <<Beetlestone >> . ERROR:parse_from_header: bad from header And I end up with a wrecked From address then nothing works. I have checked RFC 3261 and it only says quote marks should be used if the field contains a comma, question-mark or semi-colon. So it seems Linksys are following spec and SER is in error. Any comments on this? Thanks, Pat -- Patrick Jordan-Smith Network Operations Mercury Telecommunications Ltd / Quicksilver Internet voice +64-9-916-0300 / +64-27-5900595 fax +64-9-916-0301

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[Users] PIC/SER PA presence module port?
by Joachim Fabini 30 Nov '05

30 Nov '05

Hi, Did anyone succeed (or, at least, try) to port the new SER pa module (Jamey Hicks' additions for PIC/SER trial) to openSER? If yes: any hints on how to do this? If not a question to the developers: Are the module interfaces of SER and OpenSER still so close that a port of the PA module should be feasible without too much coding? There is much interest in presence around and having a decent presence module in OpenSER is a definite plus. tia, --Joachim

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[Users] How can I send radius authentication packet with openser
by Arda Tekin 30 Nov '05

30 Nov '05

Hi, I have installed openser, mysql, radiusclient-ng-0.5.2 successfully on REL3.0. openser works well with mysql. I need to send a radius authentication packet to a radius server(according to RFC2865). Packet contains base params: User-name (attr.1) $Username Password (attr.2) $Password NAS-Identifier (attr.4) (auto-generated) NAS-Port (attr.5) $uref State (attr.24) 0 Client-Port-DNIS (attr.30) NONE Caller-Id (attr.31) $calling I can not find a clear sample about radius. Which module is used for this purpose? Regards Arda

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[Serusers] serctl via unix socket?
by Cesc 30 Nov '05

30 Nov '05

Hi, We use SER+SEMS in my project. Recently, we moved the interface between them to unix sockets (from the "traditional" fifo). But now ... serctl won't work because it cannot find the ser fifo (for example, "serctl ul show"). Can the two (fifo and unix sock) cohexist? that is ... can ser open both at the same time and receive/send from/to both? Or, how can i make serctl use the socket interface? Regards, Cesc

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[Users] not going to failure_route
by Jayesh Nambiar 30 Nov '05

30 Nov '05

hi all, I was trying the following scenario: If a call comes from trusted source_ip, I rewritehost and forward them. But what I want is if the reply status is greater than 400, it should go to the failure route and try that destination. But the call gets replied as "500 service unavailable". I think, the call should go to the failure route if the status is greater than 300, is it right? Or i m making some logical error here. snippet of my cfg file: if (uri=~"^sip:00[0-9]*@") { #forward the trusted IPs without authentication strip(2); rewritehost("216.XX.XX.XX"); t_on_failure("1"); t_on_reply("2"); t_relay(); }; failure_route[1] { log(1, "trying next ip address"); prefix("00"); rewritehost("63.XX.XX.XX"); t_relay(); } onreply_route[2] { if(status=~"18[0-9]") { t_on_failure("0"); }; } pls help me out with this. thanx a lot in advance!! Jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

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sr-users November 2005