- sr-users - kamailio.org

RE: [Serusers] Packets fragmentation and message reduction
by Vitaly Nikolaev 03 Nov '05

03 Nov '05

MTU 1500 is kind a everywhere (almost) if you increase it on your interface/router then your ISP will find out that your packet too long and split it.. or block In my case I removed all information fields from SIP header -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Arek Bekiersz Sent: Thursday, November 03, 2005 4:11 AM To: Federico Giannici; SER Users Subject: Re: [Serusers] Packets fragmentation and message reduction Hi Federico, Had the same problems. Solutions that work: - you have somewhere MTU limit of 1500 bytes. Check where you have this limit, it is possible you will be able to send bigger messages after changes - use SER's consume_credentials() function to remove Authorization or Proxy-Authorization HF after use - configure all phones to use and describe in SDP only 1 or 2 codecs (decide about your system codec, delete the rest from UAs) - write your small function (in example in nathelper module) to edit SDP and remove unused codecs when messages go through proxy Arek, ----- Original Message ----- From: "Federico Giannici" <giannici(a)neomedia.it> To: "SER Users" <serusers(a)lists.iptel.org> Sent: Thursday, November 03, 2005 1:24 AM Subject: [Serusers] Packets fragmentation and message reduction > I found a situation where some SIP phones worked and some don't. > > After a while I found that, because of the added bytes of the > "Proxy-Authorization:" header, the messages generated by some sip phones > were larger than 1480 bytes and this caused the connection with the PSTN > gateway to not work. > > I tried to forwad the message with the t_relay_to_tcp() function, but by > the ngrep output it seems that it still uses UDP (but I may be wrong). > > Anyway, what can be done in this situations? > > What about a SER module that rewrite all header field names with their > compact form? Often, the message is only a few bytes more than 1480, so > this could be a solution... > > What do you think? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] mathing prefix
by Iqbal 03 Nov '05

03 Nov '05

Hi Anyone know regex to match 0061 andf not 00611, basically match australia, and not australia mobile, have thought of 0061[2-9], but if I need to do this for all countries its a pain, since they all have their own logic. iqbal

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[Serusers] Problem in SER+SEMS conference with NAT
by Ravic Costa 03 Nov '05

03 Nov '05

Hello, We have configured SER 0.9.4 and SEMS last cvs version in order to implement a conference funcionallity. Everything is working fine when the clients are using public IPs. When a client starts a conference, he listens a message that in on a wav file. When a client is behind NAT, when he tries to start a conference, he doen's listen to any message and in the soft phone (X-Lite) it appears a message saying "connected". We have captured the traffic with tcpdump, on the machine that ser and sems is running, and we've noticed that the RTP packets that the client behind NAT send are well received, however, there is only one RTP packet sent by the SER to the Client, and the Client does not get that message, maybe because he is behind NAT. Any ideia? In the ser.cfg file, to the conference funcionallity we have : _________________________________________________________ route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # select messages to redirect: if ( method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" ){ # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); # actively absorb ACKs if (method == "ACK") { t_relay(); break; } if (method=="INVITE"){ if (nat_uac_test("19")) { setflag(6); } # redirect the call to the 'conference' plug-in # if the URI begin with 100 if (uri=~"sip:100.*@") { # assumes that Sems configuration parameter 'socket_name=' # has been set to /tmp/am_sock if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems"); }; break; route(4); route(1); }; } else if (method=="BYE" || method=="CANCEL") { # Sems should already know which plug-in is handling that # call. "bye" is no plug-in name. It is a reserved name which # tells Sems to terminate the call. if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; }; } _______________________________________________________ Best regards, Ravic Costa

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[Serusers] Redirect server
by harry gaillac 03 Nov '05

03 Nov '05

Hello, I need your advises Here is my problem : I setup asterisk ser : Asterisk 5050-- |--Firewall---|----- SIP User agents SER 5060--- My client use SER as outbound proxy and asterisk as sip registrar. SER is able to forward SIP register request to asterisk however . How SER can inform SIP User agents to send register requests to asterisk. May I have to configure a redirect server to advise SIP User agents of the sip register server (asterisk)? can you help me for ser.cfg if possible ? Regards ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] accounting on mysql
by Colin Jordan 03 Nov '05

03 Nov '05

Hi there Can you please help me to setup accounting on mysql. I need to log the call transactions. Best regards, Colin

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[Serusers] problem with ser registration
by rajesh volluru 03 Nov '05

03 Nov '05

hi I get Process Request Timedout ever whenever i try to register with SER. I get following messages on starting ser. 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57' 1(0) DEBUG: init_mod_child (-1): usrloc 1(0) get_connection(): Connection not found in the pool 2(0) INFO: fifo process starting: 10317 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(10313) DEBUG: init_mod_child (1): tm 0(10313) DEBUG: callid: '3fbfd771-10313(a)69.229.23.57' 0(10313) DEBUG: init_mod_child (1): usrloc 0(10313) get_connection(): Connection not found in the pool 1(10316) DEBUG: init_mod_child (-1): auth_db 1(10316) get_connection(): Connection not found in the pool 0(10313) DEBUG: init_mod_child (1): auth_db 0(10313) get_connection(): Connection not found in the pool 1(10316) DEBUG: init_mod_child (-1): domain 1(10316) DEBUG: init_mod_child (-1): nathelper 1(10316) rtpp_test: RTP proxy found, support for it enabled 0(10313) DEBUG: init_mod_child (1): domain 0(10313) get_connection(): Connection found in the pool 0(10313) DEBUG: init_mod_child (1): nathelper 0(10313) rtpp_test: RTP proxy found, support for it enabled 2(10317) DEBUG: init_mod_child (-2): auth_db 2(10317) get_connection(): Connection not found in the pool 2(10317) DEBUG: init_mod_child (-2): domain 2(10317) DEBUG: init_mod_child (-2): nathelper 2(10317) rtpp_test: RTP proxy found, support for it enabled 2(10317) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(10317) DEBUG: register_fifo_cmd: new command (print) registered 2(10317) DEBUG: register_fifo_cmd: new command (uptime) registered 2(10317) DEBUG: register_fifo_cmd: new command (version) registered 2(10317) DEBUG: register_fifo_cmd: new command (pwd) registered 2(10317) DEBUG: register_fifo_cmd: new command (arg) registered 2(10317) DEBUG: register_fifo_cmd: new command (which) registered 2(10317) DEBUG: register_fifo_cmd: new command (ps) registered 2(10317) DEBUG: register_fifo_cmd: new command (kill) registered 2(10317) WARNING: no fifo_db_url given - fifo DB commands disabled! Can anyone please have a look at above information and let me know what is wrong with ser? Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here --------------------------------- Enjoy this Diwali with Y! India Click here

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[Serusers] SRC Port Change
by Ritesh Jalan 03 Nov '05

03 Nov '05

Hi Pls. help us in resolving the problem that the SIP proxy server changes the Source and Destination port while replying for message. Pls. find the below message for the Reference In below message A cisco ata device (which is behind NAT with IP 192.168.10.52) is trying to register it self with SIP proxy server IP 172.16.101.10 with SRC port 5060 and DST port 5060, but when SIP proxy server replies back to the message the port number gets changed i.e SRC port 53492 and DST port 5060. Is there any way by which we can fix this port num. No. Time Source Destination Protocol Info 532 152.672945 172.16.101.251 172.16.101.10 SIP Request: REGISTER sip:172.16.101.10 Frame 532 (469 bytes on wire, 469 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.280623000 Time delta from previous packet: 0.007492000 seconds Time since reference or first frame: 152.672945000 seconds Frame Number: 532 Packet Length: 469 bytes Capture Length: 469 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Cisco_b5:15:f0 (00:04:c1:b5:15:f0), Dst: Dell_17:ea:82 (00:14:22:17:ea:82) Destination: Dell_17:ea:82 (00:14:22:17:ea:82) Source: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.251 (172.16.101.251), Dst: 172.16.101.10 (172.16.101.10) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x68 (DSCP 0x1a: Assured Forwarding 31; ECN: 0x00) 0110 10.. = Differentiated Services Codepoint: Assured Forwarding 31 (0x1a) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 455 Identification: 0x01f8 (504) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 249 Protocol: UDP (0x11) Header checksum: 0x1e63 [correct] Source: 172.16.101.251 (172.16.101.251) Destination: 172.16.101.10 (172.16.101.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 435 Checksum: 0x6c94 [correct] Session Initiation Protocol Request-Line: REGISTER sip:172.16.101.10 SIP/2.0 Method: REGISTER Resent Packet: True Suspected resend of frame: 34 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK9749dfca14c0ffb2 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 Call-ID: 3431569068(a)192.168.10.52 CSeq: 1 REGISTER Contact: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 Contact Binding: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 URI: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp> SIP Display info: 23456 SIP contact address: sip:23456@192.168.10.52:5060 User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A) Content-Length: 0 No. Time Source Destination Protocol Info 533 152.674292 172.16.101.10 172.16.101.251 SIP Status: 401 Unauthorized (0 bindings) Frame 533 (432 bytes on wire, 432 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.281970000 Time delta from previous packet: 0.001347000 seconds Time since reference or first frame: 152.674292000 seconds Frame Number: 533 Packet Length: 432 bytes Capture Length: 432 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Dell_17:ea:82 (00:14:22:17:ea:82), Dst: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Destination: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Source: Dell_17:ea:82 (00:14:22:17:ea:82) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.10 (172.16.101.10), Dst: 172.16.101.251 (172.16.101.251) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3; ECN: 0x00) 0110 00.. = Differentiated Services Codepoint: Class Selector 3 (0x18) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 418 Identification: 0x0000 (0) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x9988 [correct] Source: 172.16.101.10 (172.16.101.10) Destination: 172.16.101.251 (172.16.101.251) User Datagram Protocol, Src Port: 53492 (53492), Dst Port: 5060 (5060) Source port: 53492 (53492) Destination port: 5060 (5060) Length: 398 Checksum: 0x0ad1 [correct] Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: True Suspected resend of frame: 480 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;received=172.16.101.251;branch=z9hG4bK9749dfca14c0ffb2 Call-ID: 3431569068(a)192.168.10.52 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 CSeq: 1 REGISTER WWW-Authenticate: DIGEST realm="CISCO", nonce="4369a82d", qop="auth", algorithm=MD5 Content-Length: 0 Ritesh Jalan

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[Serusers] Anyone can help me with sems
by Antonio Membrives 03 Nov '05

03 Nov '05

The scenario are ser 0.9.3 + sems 0.9.0 + mediaproxy + mysql, al instances are in the same machine The code in my ser.cfg if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { acc_db_request("Llamada perdida","missed_calls"); if(!t_write_req("/tmp/am_fifo","voicemail")){ t_reply("500","Error contacting sems voicemail"); }; #t_reply("404", "Usuario no conectado"); break; }; } else if(method=="BYE" || method == "CANCEL"){ if(!t_write_req("/tmp/am_fifo","bye")){ t_reply("500","Error contacting sems"); }; }; sl_send_reply("404", "Usuario no encontrado"); break; setflag(3); when i call a offline user, the softphone make a call but I don't ear any message or received any notification, anyone can help me? this is /var/log/messages Nov 3 11:04:56 dns2 ser[23838]: Maxfwd module- initializing Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially 110592 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially 110592 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142 Nov 3 11:04:56 dns2 ser[23847]: INFO: fifo process starting: 23847 Nov 3 11:04:56 dns2 ser[23847]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Nov 3 11:05:07 dns2 ser[23848]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:08 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:10 dns2 ser[23851]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:14 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:23 dns2 mediaproxy[3158]: command delete 3924182bdc181a37 info= Nov 3 11:05:23 dns2 mediaproxy[3158]: command execution time: 0.20 ms

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[Serusers] Accounting
by harry gaillac 03 Nov '05

03 Nov '05

Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? if acc module can't calculate acct-session-time why does this module provide radius support ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] SEMS + SER performance question
by Ryan Pagquil 03 Nov '05

03 Nov '05

Hi, How many simultaneous connection can SER + SEMS can handle? i deployed SER + SEMS in Pentium 2 300MHz with 128MB memory, how many connections can this handle? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] Ser+Media proxy
by KM 03 Nov '05

03 Nov '05

I have installed SER 0.9.3.2 and media proxy 1.2.1 on the same machine. The machine is on public network. Calls work ok except when I have a UA on a LAN behind NAT calling to a UA that is in the same LAN. In this case I get : +----------+-------------------------------+--------------------------+| username | contact | received |+----------+-------------------------------+--------------------------+| 22000001 | sip:22000001@192.168.1.7:5060 | sip:xxx.xxx.xxx.xxx:5060 || 22000002 | sip:22000002@192.168.1.6:5060 | sip:xxx.xxx.xxx.xxx:2557 |+----------+-------------------------------+--------------------------+The number that takes up port 5060 on the NAT server,which is 22000001 can call the other UA but not the other way round.22000002 can only receive calls.Please help me,KM

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[Serusers] about rtp mixing
by Jorge C 03 Nov '05

03 Nov '05

Hello everybody, I have a question related with conference servers. Do you now how is done the mixing in a Conference? Specifically, how input RTP streams are mixed into output string? Should I need to mix in a linear way, interleaving samples of different users? Thanks in advance!!

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[Serusers] Response Code 408 = "Request Timeout" or "ISUP 18 no user responding"
by Dmitry Isakbayev 03 Nov '05

03 Nov '05

Hi, I am trying to figure out how to interpret response code 408 in a failure_route block. It seem that there is a couple of different cases. If the code means that a PSTN gateway is not responding (the tm's fr_timer has expired), I want to try another PSTN gateway. On the other hand, if the code means that the user is not responding (the tm's fr_inv_timer has expired), I want to stop trying and forward the code to the UAC. How can I tell what response code 408 actually means? Thanks! Dmitry

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[Serusers] P-Asserted-Identity
by Atle Samuelsen 02 Nov '05

02 Nov '05

Hi guys, and girls, Male and Female, do anybody know if anybody has made a implimentation of 3325 for ser? or, is anybody makeing some form of support for this? -Atle

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[Serusers] SER behind a router
by Ángel Gómez 02 Nov '05

02 Nov '05

Hi, I am trilling to make SER workin conectec to internet via a router, the router only forwards the mensages to SER, the messages who go to the 5060 port. With this config i get SER working well locally, but when i try to register from the outside i can't. For more info I'm using sipsak for testing, and the version of ser is 0.9.3. Last thing: P L E A S E H E L P

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[Users] re: ISP charging for voip
by Yair Hakak 02 Nov '05

02 Nov '05

hello all, not sure if everyone here is familiar with the madison river case: http://informationweek.smallbizpipeline.com/60405214<http://www.tmcnet.com/tmcnet/articles/2005/voip-call-blocking-fcc-madison-r…> not exactly the same thing, because they were blocking instead of charging (they are a small rural CLEC who, like almost all phone companies, charge less than cost for access and make their profit on minutes, and voip takes away their minutes), but the principle is the same, and the FCC nailed them. However, meanwhile the FCC chairman has changed, and who knows if the general way of thinking is still the same... at very least, it's good precendent. as someone who works for a telecom regulator (i wont tell you which one and where) i'd be very interested in what happens around the world. are there countries where ISP's are not allowed to filter or otherwise interfere in traffic? I know there are companies that sell traffic filtering systems to ISP's so they can prioritize and/or slow the performance of certain customers, certain apps (filesharing and the like), etc. it's relatively trivial to configure such systems to introduce massive packet loss to anything on port 5060, for example, or deprioritize UDP packets, or things of that nature. -yair

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RE: [Serusers] voip charging by ISP's
by Vitaly Nikolaev 02 Nov '05

02 Nov '05

I believe It is unlawful act and there must be some regulation, I am not talking here that it will become known very well and this provider will lost some customers... and some other provider will claim that they voip friendly and get some more customer on it :-) ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Cesc Sent: Wednesday, November 02, 2005 1:32 PM To: Nils Ohlmeier Cc: serusers(a)lists.iptel.org; Roger Lewau; serusers-bounces(a)lists.iptel.org; Iqbal Subject: Re: [Serusers] voip charging by ISP's Well, i think the ISP may care if they are also providing similar bussiness. They won't probably be able to charge neither the voip user or voip provider, but they can do some harm. Identifying the rtp streams and sip signalling is not sooo difficult ... then, block a few messages, add delay and jitter to the rtp (or udp in general ...) and you've got a bad service, aparently coming from the voip provider. There is software to detect and do all the above (one was presented in VON this fall) ... There was an article in IEEE Spectrum (sept, october, nov ... can't remember). And all the tricks Nils mentions ... hey, maybe the advanced user, but my mom, no way! And then you need software that supports all those tricks ... that is even further away. In this sense, i think carriers have the winning side ... Regards, Cesc On 11/2/05, Nils Ohlmeier <lists(a)ohlmeier.org> wrote: On Wednesday 02 November 2005 14:28, Iqbal wrote: > I agree, I mean the next step would be to charge ebay and yahoo per > transaction, just wondering how easy it would be for ISP to setup a > monitor and a billing model on this I think the ISP should not care what it is transporting. The snail mail also does not look into what they are devlivering to charge you differently (except express delivery but thats another story). As soon as an ISP starts to look into the traffic from the user, the users will invent uncounted ways of fooling the ISP (use non-standard ports; use standard ports for the "wrong" service; use proxies; encrypt the traffic; and finally combine all this). All I can say to the ISP: happy debugging! And in the end I would be interested in how a court would judge about the outcoming bills (who has to prove what?) :-) Nils _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Manipulate To-Header
by Stefan Prelle 02 Nov '05

02 Nov '05

Hi all, is it possible to manipulate the From-Header (similiar to strip() and prefix()), before relaying a INVITE message? If so, how do I do this? Regards, Stefan

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[Users] voip charging by ISP's
by Iqbal 02 Nov '05

02 Nov '05

Hi i was looking into a model where ISP may charge (SBC is already thiking of charging Vonage :-)), now if an ISP wanted to charge, all they would need is to extract the INVITE, BYE and all the stuff in the middle, to do some billing per VoIP provider sending traffic on there network, is anyone aware of any sniffers which could do this not from a proxy, but from the network level at a ISP end. Iqbal

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[Users] TLS packages for debian unstable
by Bogdan-Andrei Iancu 02 Nov '05

02 Nov '05

Hi all, i just committed specs for debian to generate both tls and non-tls packages - thanks to Klaus for support ;). I also generated the packages for debian unstable - see : http://openser.org/pub/openser/latest/packages/deb-unstable/ in the next days I hope packages for different versions of debian will be uploaded. regards, bogdan

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[Serusers] Finding IP address of VoIP gateway sending the 5xx
by Ramin Nikaeen 02 Nov '05

02 Nov '05

Valued Friends, I am trying to process 5xx response codes and to resend the INVITE to an alternative VoIP gateway upon receiving an 5xx response from a VoIP gateway. So, I need to somehow figure out the IP address of the VoIP gateway from which SER receives the 5xx response. I tested and I receive 503 response codes only in onreply_route and not in failure_route at all. The 503 response does not include any "contact" fields that Would indicate the IP address of the VoIP gateway from which The 503 response is being received at SER. Do you know how I can figure out the IP address of the VoIP gateway from which SER is receiving the 503 response?! Thanks ramin

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[Serusers] SIP/2.0 477 error on sending to next hop occurred
by rajesh volluru 02 Nov '05

02 Nov '05

hi can anyone explain me why i get the error mentioned in below trace. Please let me know how can i resolve this. SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Call-ID: 21b297bc85242e507848f6c297cf4c3f(a)69.229.23.54 CSeq: 1 REGISTER From: "test11" <sip:test11@iptel.org:5060;transport=tcp>;tag=6626965 To: "test11" <sip:test11@iptel.org:5060;transport=tcp>;tag=321967ba5deb3e7f484a2df055818800-a111 Via: SIP/2.0/TCP 69.229.23.54:5060;branch=z9hG4bK67a6e74fa0f95b090b70ff5d3d969e73;rport=6719 Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=17793 req_src_ip=69.229.23.54 req_src_port=6719 in_uri=sip:69.229.23.57:5060;transport=tcp out_uri=sip:69.229.23.57:5060;transport=tcp via_cnt==1" Content-Length: 0 Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here

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[Serusers] Newbie question Best way to strip headers out of sip invites?
by Dale J. Robertson 02 Nov '05

02 Nov '05

I am looking for the best way to remove headers from INVITES sent to my PSTN gateway. The invites from my UA are too long and return '513's when they include authentication. My provider doesn't support BYOD so getting them to change maxlength is not an option. there are several headers that are sent by my ua that are not sent by the ua they have provided. when I strip the extra headers out using textops subst function it leaves an empty line which the pstn gateway doesn't seem to like very much. Is there some other method to strip headers? or maybe my regular expression syntax is wrong? the re I use to select the Date: header for removal is '^Date:.*$' replaced with // . I am testing my re's using sed. I must admit I know nearly nothing about using regular expressions. Thanks, Dale

2 1

[Serusers] Accounting
by harry gaillac 02 Nov '05

02 Nov '05

Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

3 3

[Users] Help with MySQL acc query
by Daryl Sanders 01 Nov '05

01 Nov '05

I'm trying to come up with a query on the acc table that will return all INVITES that don't have a matching BYE. I'm basically looking for sessions that are currently open. - Daryl

6 11

[Users] Upgrading to 1.0.0
by Brandon Price 01 Nov '05

01 Nov '05

After upgrading from 0.9.5 to 1.0.0, openser won't start and I get a bunch of "!!!attention: from v1.0.0+ use 'return' instead of 'break'" in /var/log/messages. I figured, ok must be a syntax change so I replaced all instances of break; with return; and tried again. The messages persist. I have included sample messages and corresponding block of code. Thx in advance. Nov 1 10:43:42 ser1 openser: parse error (291,1-2): overwritting request routing table Nov 1 10:47:05 ser1 last message repeated 2 times line 291 is the closing } for route block 0. The same openser.cfg starts and works with 0.9.5. what the heck?

3 4

[Users] RE: [Serusers] voip charging by ISP's
by Vitaly Nikolaev 01 Nov '05

01 Nov '05

Well, I do not see a problem all depend on kind of network that you use, for example if ISP use cisco switches they can assign one of the ports on it as MONITOR for the port with interesting data (Vonage uplink :)) Then you connect something like call analyzer or simple unix server with ngrep running :) and make some nice filter that will catch all invites/byes that you need, or may be not invites but actually detect voip rtp streams and log it Ngrep can easily be changed to dump all SIP requests into database and then you can do what you want with it. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On Behalf Of Iqbal Sent: Tuesday, November 01, 2005 12:55 PM To: serusers(a)iptel.org Cc: users openser.org Subject: [Serusers] voip charging by ISP's Hi i was looking into a model where ISP may charge (SBC is already thiking of charging Vonage :-)), now if an ISP wanted to charge, all they would need is to extract the INVITE, BYE and all the stuff in the middle, to do some billing per VoIP provider sending traffic on there network, is anyone aware of any sniffers which could do this not from a proxy, but from the network level at a ISP end. Iqbal _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] RE: [Asterisk-Users] chan_exosip2
by harry gaillac 01 Nov '05

01 Nov '05

Thanks for reply, Does Chan_exosip2 is stable where may i find help ? Chan_sip from asterisk doesn't support IM and presence via SIMPLE. I whish to use either asterisk or openpbx to provide telephony features with SER to relay IM and presence SIMPLE openpbx or asterisk | | ----SER--- | | Sip agent sip agent What do you advise me ? Regards Harry --- Joshua Colp - Asterlink <joshnet(a)nbnet.nb.ca> a écrit : > Hello Harry, > > This is rather the wrong list to ask this... since > this is Asterisk, not > OpenPBX.org > > Chan_exosip2 though is something I'm basically > designing to have 3 operating > modes. > > Full server: Most closely resembles chan_sip in that > it acts as a B2BUA > Partial proxy: Extensions are mapped to SIP URIs and > it acts as a proxy. > Gateway: No authentication occurs (this is > presumably done outside by a SIP > proxy), incoming calls just get thrown into a > context. Outgoing calls are > down via SIP URI. > > Joshua Colp > > -----Original Message----- > From: asterisk-users-bounces(a)lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On > Behalf Of harry gaillac > Sent: Tuesday, November 01, 2005 10:19 AM > To: asterisk-users(a)lists.digium.com > Subject: [Asterisk-Users] chan_exosip2 > > Hello, > > I read roadmap on www.openpbx.org. > Does chan_exosip2 will be able to provide a real sip > proxy ? > What about asterisk solutions ? > > Harry > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users(a)lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users(a)lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] dev/null
by Iqbal 01 Nov '05

01 Nov '05

Hi Anyone know howto /dev/null inbound requests, from users. Reason for this is that I have a few random setups, which I no longer support, and the user has xlite and is just sending through REGISTER requests, I'd much rather they just get dumped, prefer to do it within ser, rather than some iptables/fw block. I guess I could have some blacklist table, and if From= blacklist then break...suggestions. Iqbal

3 3

[Serusers] UA's under same NAT
by deviator 01 Nov '05

01 Nov '05

Hello! i have 0.9.5 openser + mediaproxy 1.4.2 + serweb + mysql and so on, all works fine. private and public ua's works, no problems. But i have one question. Is it possible to send rtp traffic directly to each other if a callee and a caller are under same nat ? not through mediaproxy. p.s. huge thanks to the developers of ser/openser. Nice software! RESPECT!

2 1

[Serusers] Serweb not login
by Toygun Mavinil 01 Nov '05

01 Nov '05

Hi, İ cannot use serweb, it does not work My system is: Suse 9.3 Apache 2 Php 4 Ser 0.9.4 with mysql support Serweb 0.9.4 Ser Works with mysql support but serweb does not The login screen comes to the browser and if i use wrong user/pass it gives out wrong username or password but İf i use correct user/pass as admin - heslo it gives no errors and does nothing. Login screen comes again without any error First it worked for 5 minutes i could login but then i could not do anything else because it said "session expired, please login again" i can not also use serweb logging, pear log is not installed as pear package is installed Toygun Mavinil Mavinil Bilgisayar Sistemleri Toptan ve Perakende Satış Tel: 0 212 873 97 00 Faks: 0 212 873 97 03 Adres: Adakent Cd. No:28 Ak Apt. D.16 Beylikdüzü İstanbul Web: <http://www.mavinil.com.tr> www.mavinil.com.tr <http://www.adslbilgi.com> www.adslbilgi.com

1 0

[Users] MySQL 2-way replication with SER
by Daryl Sanders 01 Nov '05

01 Nov '05

Is anyone using MySQL replication 2-way replication (dual-master, multi-master) in your SER/OpenSER setup. How well does it work? Any pitfalls? - Daryl

3 2

RE: [Users] "uri == myself" matches when port is not port is different
by Vitaly Nikolaev 01 Nov '05

01 Nov '05

I also went thought the same situation and found the best solution (fastest) to keep second application (in my case b2bua) on different IP (alias) -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Tavis P Sent: Monday, October 31, 2005 7:34 PM To: users(a)openser.org Subject: [Users] "uri == myself" matches when port is not port is different This is a somewhat unique situation, but with my config i have some call features that are hosted on another SIP aware application running on the same system. In a branch_route block i am checking to see if destination is myself to avoid loops, however when i direct a message to, say "127.0.0.1:5090" and OpenSER is configured to listen on port 5060 only the "if (uri == mysql)" statment matches even though the port is different. It seems to me that this behaviour is not intuative given the use of the statment, can anyone else comment on this? _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Users] Tracker system
by Bogdan-Andrei Iancu 01 Nov '05

01 Nov '05

Hi everybody, I just finished configuring and testing the project tracker hosted on SorceForge. My apologies for the "un-explained" emails on the devel list :) The purpose of the tracker will be of course to have a centralized system to keep trace of : 1) bug reports 2) support requests 3) feature requests 4) documentation reports 5) patch submission If needed, other sections can be added. All developer have admin rights on this tracker (to create new categories or groups, etc). let's use it and start registering all known pending issues (discussed so far on the mailing list) in order to avoid forgetting them and to have a big picture of the current status. All changes regarding the status of the register items will be emailed to the devel mailing list. There are couple of items already registered by me - all project developers: please review them and update them accordingly if necessary! For example, for the enum patch, I think Juha will be the best person to assign it to. regards, bogdan

1 1

Re: [Users] Detecting runaway calls
by Daryl Sanders 01 Nov '05

01 Nov '05

Thanks for the info guys! It sounds like I need to do a little reading up on cseq to determine if this will even work, or find a PSTN gateway provider that supports Session-Timers. - Daryl On 10/27/05, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > by sending re-INVITEs from the middle of the path you will increase the > cseq number differently on each side...so you will need to synchronize > the cseq value when some in-the-dialog requests are passing through your > proxy....and that's quite complicated.... > > regards, > bogdan > > Daryl Sanders wrote: > > >Would it be possible to fake a REINVITE then check the response to > >determine if a call is still in session? Just brainstorming... > > > >- Daryl > > > > > >

7 16

[Users] lcr module
by Richard Z 01 Nov '05

01 Nov '05

Hi, I am trying to develop some features which requires to call load_gws and next_gw in a failure route. For example, in normal routing block, it forwards the call to a subscriber's ip phone locations. If nobody answer, it tries to call a pstn number. In the failure route, i called load_gws and next_gw. However it generates an error "next_gw(): No ruri_user AVP". Looking at the code, there seems to be an assumption that first time call to next_gw always happens in the normal routing block. But in my case, it calls next_gw first time in a failure route. I even tried to manually add the new uri's username to avp i:1402. It still won't work. Is it possible to call next_gw first time in a failure route? Thanks, Richard

2 3

[Users] Call not torn down when CANCEL sent before INVITE message is finished being processed
by Tavis P 01 Nov '05

01 Nov '05

With the latest CVS build of OSER, when an INVITE is sent followed by a CANCEL, before OSER has finished processing the initial INVITE, the outbound call is not torn down, and SER routes the CANCEL message to itself. Below is an ngrep trace of the exchange. I have a development cluster of mediaproxy servers setup and one of them was taken offline but left in the loop which increased the time it took for a message to be processed which is how i came across this situation (it took some seconds before the INVITE was able to be transmitted to the nexthop_proxy) -- # U 2005/10/04 01:44:43.134010 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-34d67b8f. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 101 INVITE. Max-Forwards: 70. Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:43.135736 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-34d67b8f;rport=5070;received=SIP_UAC. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>;tag=688bd636d693eb2ab143ccbaf0db2c7a.8474. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="hq.galnet.ca", nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e", qop="auth". Content-Length: 0. . # U 2005/10/04 01:44:43.147960 SIP_UAC:5070 -> OSER_PROXY:15061 ACK sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-34d67b8f. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>;tag=688bd636d693eb2ab143ccbaf0db2c7a.8474. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 101 ACK. Max-Forwards: 70. Contact: tavis <sip:200110005334@192.168.99.66:5070>. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 0. . # U 2005/10/04 01:44:43.154728 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="6867ef34181ba7cedbcbf8dd869f53ab",qop=auth,nc=00000001,cnonce="750e43e5". Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:43.648366 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="6867ef34181ba7cedbcbf8dd869f53ab",qop=auth,nc=00000001,cnonce="750e43e5". Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:44.155113 SIP_UAC:5070 -> OSER_PROXY:15061 CANCEL sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 CANCEL. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="512097281b4fba990adf2cc6c273f033",qop=auth,nc=00000002,cnonce="750e43e5". User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 0. . # U 2005/10/04 01:44:44.285994 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 483 Too many hops. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>;tag=688bd636d693eb2ab143ccbaf0db2c7a.32f9. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 CANCEL. Content-Length: 0. . # U 2005/10/04 01:44:44.648305 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="11a15f9ef4fdb1c9ab05566cad92d67e",qop=auth,nc=00000002,cnonce="750e43e5". Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:44.688471 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787;rport=5070;received=SIP_UAC. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Content-Length: 0. . # U 2005/10/04 01:44:44.689012 OSER_PROXY:15061 -> NEXTHOP_PROXY:5060 INVITE sip:16046946333@NEXTHOP_PROXY SIP/2.0. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bK11ea.6c311116.0. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 15. Contact: tavis <sip:200110005334@SIP_UAC:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 288. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:45.396833 OSER_PROXY:15061 -> NEXTHOP_PROXY:5060 INVITE sip:16046946333@NEXTHOP_PROXY SIP/2.0. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bK11ea.6c311116.0. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 15. Contact: tavis <sip:200110005334@SIP_UAC:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 288. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:46.090526 NEXTHOP_PROXY:5060 -> OSER_PROXY:15061 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bK11ea.6c311116.0. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 174. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 NEXTHOP_PROXY. t=0 0. m=audio 19004 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. # U 2005/10/04 01:44:46.104387 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 176. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. # U 2005/10/04 01:44:46.190376 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 176. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. ## U 2005/10/04 01:44:46.702880 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 176. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000.

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Re: [Users] Insert name into from header of SIP messages
by Brandon Price 01 Nov '05

01 Nov '05

Following that logic should I not be able to just replace the display and leave the uri as is with... uac_replace_from("$avp(s:cnam)",""); I have of coarse already trid that with no success. On Oct 28, 2005, at 3:38 PM, Norman Brandinger wrote: > If you break your callerid into 2 parts, you should be able to use > something like the following: > > AVP display: "John Doe" > AVP uri: "sip:18005551212@example.com" > > uac_replace_from("$avp(s:display)","$avp(s:uri)"); > > The above information has been documented at http://www.openser.org/ > docs/modules/1.0.x/uac.html > > Regards, > Norm > > Brandon Price wrote: > >> We are storing cnam (callerid name) values in a mysql database for >> use with openser. I have openser pulling the values from the >> database using avpops but I need to know how to insert the value >> into the from so that it will be passed to my sip user agents. >> Related code posed below >> >> modparam >> ("avpops","db_scheme","cnam_scheme:username_col=phonenumber;table=cna >> m_cache;value_col=cnam;value_type=string") >> modparam("avpops", "avp_aliases", "cnam=s:cnam") >> >> avp_db_load("$from/cnam_scheme","cnam/$cnam_scheme"); #sample >> query -- select cnam from cnam_cache where phonenumber='13143212222' >> >> this should now store that value in $cnam. >> now how do I put that value for example "John Smith" in the from >> portion of sip messaging so that it displays on the users sip >> phone, or phone attached to an ata? >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >

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[Serusers] Fwd: DIDX.org - DIDXchange
by Rehan Ahmed 01 Nov '05

01 Nov '05

---------- Forwarded message ---------- From: Rehan Ahmed <rehan33(a)gmail.com> Date: Nov 1, 2005 8:02 AM Subject: DIDX.org - DIDXchange To: serusers(a)lists.iptel.org Hello All, I wanted to know if someone here is using did numbers from DIDX.org and able to make them work on SER yet. Rehan

1 0

[Serusers] when I include mysql module in ser.cfg, UA receive "483, too many hops", any advice?
by fishman7788 01 Nov '05

01 Nov '05

I have use ser_mysql create to setup database,and can connect to ser,can add a user using serctl add,but UA just can't receive correct register message,any suggestion? Thanks regargs free

1 0

[Serusers] ser.cfg parse error
by Ma Zhiyong 01 Nov '05

01 Nov '05

Hi All, After I installed the latest CVS version, I found SER could't be started by serctl utility. It shows # serctl start Starting SER : 0(14877) read 3820494812 from /dev/urandom 0(14877) seeding PRNG with 656407867 0(14877) test random number 38195676 # 0(14877) parse error (1,1-2): parse error .......... It seems that my ser.cfg file has wrong format or other problem. Any one has the same matther before? Does that mean my ser.cfg file encounters "parse error" at line 1 and coloum 1 to 2? Thanks.

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Re: [Serusers] Error with acc.so and db funtions
by zhangshuai 01 Nov '05

01 Nov '05

Dear Antonio Membrives, Did you see the makefile of acc module and recompile acc.so? >Hi again i have this lines in ser.cfg > >modparam("acc", "log_flag", 1) >modparam("acc", "log_level", 2) >modparam("acc", "db_flag", 1) >modparam("acc", "db_missed_flag", 3) >modparam("acc", "log_missed_flag", 3) >modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") >modparam("acc", "report_ack", 0) >modparam("acc", "log_fmt", "miocfsp") > >for store missing calls in serweb, but when I launch the ser services >received 3 errors and see this in /var/log/messages > >Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_flag> not found >in module <acc> >Oct 31 13:50:00 dns2 ser: parse error (73,19-20): Can't set module parameter > >Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_missed_flag> >not found in module <acc> >Oct 31 13:50:00 dns2 ser: parse error (74,19-20): Can't set module parameter > >Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_url> not found >in module <acc> >Oct 31 13:50:00 dns2 ser: parse error (76,20-21): Can't set module parameter > >I have fedora 3+ser 0.9.3 , any idea of this. > >Thanks. > >Antonio Membrives > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > = = = = = = = = = = = = = = = = = = = =

1 0

[Users] "uri == myself" matches when port is not port is different
by Tavis P 01 Nov '05

01 Nov '05

This is a somewhat unique situation, but with my config i have some call features that are hosted on another SIP aware application running on the same system. In a branch_route block i am checking to see if destination is myself to avoid loops, however when i direct a message to, say "127.0.0.1:5090" and OpenSER is configured to listen on port 5060 only the "if (uri == mysql)" statment matches even though the port is different. It seems to me that this behaviour is not intuative given the use of the statment, can anyone else comment on this?

1 0

[Serusers] Please note the "HOW TO" for MYSQL connect issue
by Dhiman D Chowdhury 31 Oct '05

31 Oct '05

Dear friends, If you are like me faced with an issue describe below and could not find the answer for it. Please note the recommendation below. setup: Redhat Linux 9.0 and Mysql 3.23 Issue: 1. while trying to connect from shell prompt; #mysql Eorr 1045: Access denied for root@localhost (using password: No). Note: you will see this message in different times perhaps when using: # mysql.sh create or similar commands. Recommendation: There is a bug in Mysql version 3.23: Please visit the url below: http://bugs.mysql.com/bug.php?id=11822 To overcome the issue: Simply use the following command: mysql -h 127.0.0.1 -u root -p [issue it as needed] Thanks Dhiman

1 0

[Users] problems with mediaproxy
by Gustavo Hernandez Baratta 31 Oct '05

31 Oct '05

Hi all! I´m trying to setup mediaproxy but making calls this error occur with proxydispatcher: Starting SER MediaProxy server: mediaproxyListening for commands on local socket `/var/run/proxydispatcher.sock' Default mediaproxy server is on `XX.XXX.XXX.XXX:25060' : : ---------------------------------------- Exception happened during processing of request from Traceback (most recent call last): File "/usr/lib/python2.2/SocketServer.py", line 458, in process_request_thread self.finish_request(request, client_address) File "/usr/lib/python2.2/SocketServer.py", line 253, in finish_request self.RequestHandlerClass(request, client_address, self) File "/usr/lib/python2.2/SocketServer.py", line 514, in __init__ self.handle() File "/usr/local/mediaproxy/modules/dispatcher.py", line 303, in handle result = self.execute(command) File "/usr/local/mediaproxy/modules/dispatcher.py", line 347, in execute proxy = MediaProxy('domain://%s' % domain).address File "/usr/local/mediaproxy/modules/dispatcher.py", line 578, in __init__ self.address = proxy File "/usr/local/mediaproxy/modules/dispatcher.py", line 583, in __setattr__ if name in readonly and self.__dict__.has_key(name): TypeError: 'in <string>' requires character as left operand ---------------------------------------- : ---------------------------------------- Exception happened during processing of request from Traceback (most recent call last): File "/usr/lib/python2.2/SocketServer.py", line 458, in process_request_thread self.finish_request(request, client_address) File "/usr/lib/python2.2/SocketServer.py", line 253, in finish_request self.RequestHandlerClass(request, client_address, self) File "/usr/lib/python2.2/SocketServer.py", line 514, in __init__ self.handle() File "/usr/local/mediaproxy/modules/dispatcher.py", line 303, in handle result = self.execute(command) File "/usr/local/mediaproxy/modules/dispatcher.py", line 347, in execute proxy = MediaProxy('domain://%s' % domain).address File "/usr/local/mediaproxy/modules/dispatcher.py", line 578, in __init__ self.address = proxy File "/usr/local/mediaproxy/modules/dispatcher.py", line 583, in __setattr__ if name in readonly and self.__dict__.has_key(name): TypeError: 'in <string>' requires character as left operand ---------------------------------------- Ideas? Thanks a lot, gus

1 0

[Users] version 1.0.0 released
by Daniel-Constantin Mierla 31 Oct '05

31 Oct '05

Hi everybody, the version 1.0.0 has been released. The branch 'rel_1_0_0' is created and must be used by anybody wanting the get v1.0.0 out from CVS (cvs co -r rel_1_0_0 sip-server). There are a lot of new features in this release, starting from TLS, pseudo-variables in core and continuing with enhancements in avpops and other module. This release is the first open source sip server with TLS support, although there is still to work on it (e.g., multi domain support) it allow secure sip signaling communications between servers and clients. An unified format of pseudo-variables has been adopted in core and many modules, basic functionality for future dialog support is present in RR module, avpops includes functions for string operations (concatenation, prefix, suffix, substitute), arithmetic and bitwise operations, among others. A new module, uac_redirect, has been introduced to handle redirect replies (3xx) on server, with proper accounting and filter rules. See the full list of new features at: http://www.openser.org/release-1.0.0.php A roughly change log can be browsed online at: http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/ChangeLog?view=mar… There are two categories of released packages: - classic openser, without TLS support - the name of the package does not contain the -tls- word - TLS-ed version of openser - the name of the package contain the -tls- word The tarball with the sources can be downloaded from: http://openser.org/pub/openser/1.0.0/src/ The basic binary packages are deployed to the download site: http://openser.org/pub/openser/1.0.0/bin/ http://openser.org/pub/openser/1.0.0/packages/ We are still collecting other packages from contributors which already volunteered, so expect new distributions to be represented there in the next days. The module documentations are up-to-date, the site and tutorials need a quick review in the next days.

4 4

[Serusers] Too many hops
by Philip Jones 31 Oct '05

31 Oct '05

I am using a dedicated server to host SER, when I use the actual IP address of server i/f I connect no problem but, using the domain name, the client times out with "Too many hops". Is there a known solution to this? I would really prefer to use the domain name. Phil

1 0

[Serusers] REFER with Replaces header via FIFO - Invite drops Replaces
by Brian Roy 31 Oct '05

31 Oct '05

Short and sweet: I am attempting to REFER and existing call to another UA using a REFER with an imbedded REPLACES header in the REFER-TO header. FIFO accepts the REFER and the call is set up, however the REPLACES header is dropped between the REFER and the INVITE. 1) Has anyone been able to get a REFER with REPLACES to work with SER? a. Am I doing something completely friggin' stupid?? 2) Is this a SER bug. or a FIFO bug. or neither? See Below: Initial Call: *************************You'll note that replace is supported*************************************** INVITE sip:1999206@maggie-may.home.brianandkelly.ws:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bK5d9b3bcbFDFC6408 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060> CSeq: 1 INVITE Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130777573 1130777573 IN IP4 192.168.0.161 s=Polycom IP Phone c=IN IP4 192.168.0.161 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 ACK sip:206@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKeb569c391EDD1A0 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060>;tag=as1624b103 Route: <sip:192.168.0.24;ftag=2A63079D-99C40A74;lr=on> CSeq: 1 ACK Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Max-Forwards: 70 Content-Length: 0 Attempted REFER: ***************************************Note that that Replaces info drops in the INVITE*********************************************************** REFER sip:kellydev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.24;branch=z9hG4bK3915.d2aefa37.0 To: sip:briandev@maggie-may.home.brianandkelly.ws From: sip:kellydev@maggie-may.home.brianandkelly.ws;tag=1f850894c564e190ae8e24 413352beae-4b42 CSeq: 10 REFER Call-ID: 2bbbc774-0(a)192.168.0.24 Content-Length: 0 User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Refer-To: sip:briandev@maggie-may.home.brianandkelly.ws?Replaces=74e15821-cc9958e7 -2e639266(a)192.168.0.161;to-tag=as1624b103;from-tag=2A63079D-99C40A74 INVITE sip:briandev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK880b0a28917F93C5 From: "Kelly Roy - DEV" <sip:kellydev@maggie-may.home.brianandkelly.ws:5060>;tag=519BC0C2-649892 E1 To: <sip:briandev@maggie-may.home.brianandkelly.ws> CSeq: 1 INVITE Call-ID: 801c0ab6-44e87514-ba253cb(a)192.168.0.160 Contact: <sip:kellydev@192.168.0.160:5060;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130775178 1130775178 IN IP4 192.168.0.160 s=Polycom IP Phone c=IN IP4 192.168.0.160 t=0 0 m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685) Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685)

1 0

[Serusers] REFER with Replaces header via FIFO - Invite drops Replaces
by Brian Roy 31 Oct '05

31 Oct '05

Short and sweet: I am attempting to REFER and existing call to another UA using a REFER with an imbedded REPLACES header in the REFER-TO header. FIFO accepts the REFER and the call is set up, however the REPLACES header is dropped between the REFER and the INVITE. 1) Has anyone been able to get a REFER with REPLACES to work with SER? a. Am I doing something completely friggin' stupid?? 2) Is this a SER bug. or a FIFO bug. or neither? See Below: Initial Call: *************************You'll note that replace is supported*************************************** INVITE sip:1999206@maggie-may.home.brianandkelly.ws:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bK5d9b3bcbFDFC6408 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060> CSeq: 1 INVITE Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130777573 1130777573 IN IP4 192.168.0.161 s=Polycom IP Phone c=IN IP4 192.168.0.161 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 ACK sip:206@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKeb569c391EDD1A0 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060>;tag=as1624b103 Route: <sip:192.168.0.24;ftag=2A63079D-99C40A74;lr=on> CSeq: 1 ACK Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Max-Forwards: 70 Content-Length: 0 Attempted REFER: ***************************************Note that that Replaces info drops in the INVITE*********************************************************** REFER sip:kellydev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.24;branch=z9hG4bK3915.d2aefa37.0 To: sip:briandev@maggie-may.home.brianandkelly.ws From: sip:kellydev@maggie-may.home.brianandkelly.ws;tag=1f850894c564e190ae8e24 413352beae-4b42 CSeq: 10 REFER Call-ID: 2bbbc774-0(a)192.168.0.24 Content-Length: 0 User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Refer-To: sip:briandev@maggie-may.home.brianandkelly.ws?Replaces=74e15821-cc9958e7 -2e639266(a)192.168.0.161;to-tag=as1624b103;from-tag=2A63079D-99C40A74 INVITE sip:briandev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK880b0a28917F93C5 From: "Kelly Roy - DEV" <sip:kellydev@maggie-may.home.brianandkelly.ws:5060>;tag=519BC0C2-649892 E1 To: <sip:briandev@maggie-may.home.brianandkelly.ws> CSeq: 1 INVITE Call-ID: 801c0ab6-44e87514-ba253cb(a)192.168.0.160 Contact: <sip:kellydev@192.168.0.160:5060;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130775178 1130775178 IN IP4 192.168.0.160 s=Polycom IP Phone c=IN IP4 192.168.0.160 t=0 0 m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685)

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[Serusers] Windows Messenger 5.1 On-line
by gorand＠dvvti.com 31 Oct '05

31 Oct '05

I am testing Windows Messenger 5.l and it works fine except when I connect other users that are on the SER box, it does not show them as on-line. It displays them as off-line. I am able to make sip calls through the SER Box. Everything works fine in that aspect. I want to use the Windows Messenger as a communication aspect to be able to send instant messages to the users. Please let me know if this possible with this Windows Messenger? Thanks.

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[Users] How to remove fields from SQL accounting?
by Federico Giannici 31 Oct '05

31 Oct '05

I need to customize the data logged with the SQL accounting. The "db_extra" parameter let me ADD new fields but it seems that there is no way to REMOVE fields from the sql command. Infact, the "log_fmt" seems to affect only the syslog accounting. Is there a way to prevent a field to be included into the SQL INSERT command? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] Handling of REFER messages
by Miroslaw Nawrot 31 Oct '05

31 Oct '05

Hi, I am trying to do a peculiar task and I want to find out if SER can do it. The issue is with call transfers which generate the REFER message. I am connecting a Nortel voip switch to Asterisk via sip. I need to have all call transfers generated by the Nortel switch to not reach Asterisk as all the call transfers are done to dids/extensions not found on the Asterisk PBX. Instead I would like to intercept / re-route them back to my Nortel switch which can handle it. Currently I have a workaround using an Audiocodes gateway with loopback cables. The Audiocodes when sees the transfer request sends it back to the Nortel switch. As this is a wasteful use of expensive hardware I would like to find another method to do this. Can someone let me know if this is possible with SER? And if it is which functions do I use? Any help is greatly appreciated. Mirek The Future of Voice Globalive Communications Corp. T: 416-640-1088 F: 416-640-1089 60 Adelaide St. E, 6th Floor Toronto, ON, M5C 3E4 www.globalive.com

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[Serusers] Error with acc.so and db funtions
by Antonio Membrives 31 Oct '05

31 Oct '05

Hi again i have this lines in ser.cfg modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsp") for store missing calls in serweb, but when I launch the ser services received 3 errors and see this in /var/log/messages Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_flag> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (73,19-20): Can't set module parameter Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_missed_flag> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (74,19-20): Can't set module parameter Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_url> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (76,20-21): Can't set module parameter I have fedora 3+ser 0.9.3 , any idea of this. Thanks. Antonio Membrives

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[Serusers] Error with msilo, dont work
by Antonio Membrives 31 Oct '05

31 Oct '05

Hi to all, I try to use MSILO in Fedora 3 + Ser 0.9.3 and received the next message in /var/log/messages /var/log/tail messages Oct 31 09:54:15 dns2 ser: WARNING: fix_socket_list: could not rev. resolve x.x.x.x Oct 31 09:54:15 dns2 ser[8823]: Maxfwd module- initializing Oct 31 09:54:15 dns2 ser[8823]: bind_dbmod: Module mysql//ser does not export db_use_table function Oct 31 09:54:15 dns2 ser[8823]: init_mod(): Error while initializing module msilo And ser not work, if I coment the line of msilo db_url work again but not work msilo. Any idea? Antonio Membrives. #ser.cfg debug=3 fork=yes log_stderror=no listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE alias=dominio.es alias=x.x.x.x port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registar@telintcable.es") modparam("msilo", "db_url", "mysql//ser:heslo@localhost/msilo") #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 2) #modparam("acc", "log_missed_flag", 3) #modparam("acc", "log_fmt", "fimos") #modparam("acc.", "db_url","mysql://ser:heslo@localhost/ser") #modparam("acc", "db_flag", 2) #modparam("acc", "db_missed_flag", 3) modparam("tm","fr_inv_timer",30) modparam("tm","fr_timer",3) /var/log/tail messages Oct 31 09:54:15 dns2 ser: WARNING: fix_socket_list: could not rev. resolve x.x.x.x Oct 31 09:54:15 dns2 ser[8823]: Maxfwd module- initializing Oct 31 09:54:15 dns2 ser[8823]: bind_dbmod: Module mysql//ser does not export db_use_table function Oct 31 09:54:15 dns2 ser[8823]: init_mod(): Error while initializing module msilo

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[Serusers] where to find serweb tool
by dave cra 31 Oct '05

31 Oct '05

Hi all, I've tried to get the serweb tool off the iptel website and it can not find it. Is there another place to obtain this? I'm running windows 2000. Cheers Dave --------------------------------- Yahoo! FareChase - Search multiple travel sites in one click.

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[Users] pa module
by Schweizer Laurent 31 Oct '05

31 Oct '05

Hello, I try to setup the pa module and I have some errors. The pa always send a NOTIFY with a status set to CLOSED. I have also see in some olds mails that it is possible to add an xcap server, what is doing this server? Is a free xcap server available? Thanks Laurent My config: modparam("pa", "default_expires", 1800) modparam("pa", "timer_interval", 10) modparam("pa", "use_db", 0) # def=0 disable db support modparam("pa", "use_place_table", 1) # def=0 disable use of place table modparam("pa", "use_bsearch", 0) # def=0 disable use of binary search for location modparam("pa", "use_location_package", 0) # def=0 disable use of location package #modparam("pa", "pa_domain", "bigU.edu") # def name of xcap server modparam("pa", "db_url", "mysql://..........") modparam("pa", "presentity_table", "presentity") modparam("pa", "presentity_contact_table", "presentity_contact") modparam("pa", "watcherinfo_table", "watcherinfo") modparam("pa", "place_table", "place") modparam("pa", "new_watcher_pending", 0) # def=0 default state ofwatchers 0=active modparam("pa", "pidf_priority", 1) # def=1 default priority of contacts in pidf documents

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[Serusers] fee-for-service SER server - clarified
by Michael Kendall 31 Oct '05

31 Oct '05

Greger, Thanks so much for your reply to my question, and thanks also to Olivier, Bravo, Iqbal, and Phil, who each sent me some great SER server recommendations or requested clarification to my confusing question. You are right that I did not provide enough info in my question. I do not need any of the example services that you mentioned in a, b, c, d. I don't want to control or customize any SER server, I only want to be a normal customer of an existing SER server. I do not need any access to regular telephones. What I want to do is have a call center with 10 tutors, with each tutor connected to a different remote student, The 10 teachers and 10 students would each be using the Windows (SIP) Messenger softphone and would find and call each other using the SER server. The SER server would need to allow media-proxy service so that we can traverse NATs and firewalls. I have selected Windows (SIP) Messenger as my softphone because it is cost-free (quite important if I have 300 students) and because it has a whiteboard (required for the tutoring). If someone can suggest a better softphone which has a whiteboard I would be very happy, especially if the softphone supports STUN. If the suggested softphone is free, that would be perfect. I can't really afford to use a softphone which charges me a fee for each named user, but if I need to pay a reasonanable price for each SIMULTANEOUS user, that would be ok. Most of the SER server websites that I have visited seem to recommend the Xten softphone, but since it has no whiteboard, I cannot use it. Windows (SIP) Messenger does not appear to support STUN, so media-proxy is the only way I know to traverse NATs. (Please correct me if I am wrong about Messenger not supporting STUN.) I know that the newest versions of Windows (SIP) Messenger do not allow connecting to 3rd party servers, so I plan to use the older versions that do allow it. Some of the SER server websites ( like fonosip ) claim (after a search) that they support Windows (SIP) Messenger, but they do not offer any configuration instructions. The fonosip site also does not say whether they will offer media-proxy services, which I need for NAT traversal. It is strange that these SER service provider websites provide very little info about whether they support media-proxy or how to configure a softphone for STUN. I know that media-proxy places a heavy load on the SER server, and that some SER servers are therefore reluctant to consume their expensive bandwidth by offering media-proxy services. But since there must be many other customers who are willing to pay for media-proxy service, there should be a group of SER servers who are offering this service for a fee. I just need to find one. Can someone please suggest a SER server which might satisfy the Messenger and media-proxy requirements? And can someone recommend a better softphone with whiteboard? thanks, Michael Greger wrote: Dear Michael, I know that many on this list operate own services they sell in the market. Others sell software packages and services, both to corporations (ex. Asterisk + ser in a combo setup) and to service providers (some sort of white-label service). There are also hosted services for enterprises that are not based on SER. I would suspect that lack of answer may have something to do with lack of information. You are not saying what you need. Here are some examples: a. A hosted SER server where you can log in and do changes to ser.cfg (for fun or commercial) b. A hosted corporate PBX-type solution c. A hosted SIP service ? la university setups with ENUM lookup and authentication against your own user database d. A white-label service that you will brand as your own and resell in a given market And BTW, you are not saying anything about your requirements (just a vanilla SER server?!) nor the size of the subscriber base you want to support etc etc. If you are looking for b, you are probably better off looking in your local market for a telephony provider who can give you IP access. g-)

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[Serusers] need a help!!!
by deviator 31 Oct '05

31 Oct '05

Hello! I'm using openser 0.9.5 with rtpproxy, i'm found a good config for this on onsip.org site. All works fine with public ips ua's, but i have a problem with nat :( server have a public ip, ua's have a private ips, symmetric nat. I have an acknowledgement error (ACK Timeout) and calls failure after 30-32 sec. What should i change or add in openser.cfg (nat-rtpproxy.5.0.cfg version) pls help!!!

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[Serusers] When to use MediaProxy
by tcchan 31 Oct '05

31 Oct '05

Hi All, I have SER work together with MediaProxy and two UAs; UA1 is NATED whereas UA2 has public IP. I have something like this to process the INVITE if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; t_on_reply("1"); and on the reply_route, I have if (client_nat_test("1")) { setflag(4); }; if (isflagset(4) && status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { fix_contact(); use_media_proxy(); } }; Now, if UA1 (nated) calls UA2, it is OK ( mediaproxy is properly setup and both side can hear each other. The client_nat_test() is met and flag4 is set which caused both the invite and the on_reply to use_media_proxy). If UA2(public) calls UA1 (nated), problem arises, UA2 can hear UA1 but not the reverse. The client_nat_test("3") is not met and thus flag4 not set. On the reply_route, although client_nat_test("1") is met, flag4 set and use_media_proxy() is executed, it return error: use_media_proxy(): empty response from mediaproxy ERROR: on_reply processing failed To temporary work around this problem, I forcefully set flag4 on and thus all calls route through the mediaproxy ( which is expensive in bandwidth usage). Is there any better way to solve this? Any suggestion? Has anybody already looks into this and has a solution? Thanks for any help available. Regards, TC Chan

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[Serusers] Inserting SER into my voice network
by Ray Van Dolson 31 Oct '05

31 Oct '05

First off, here is my voice network layout currently: http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png We're using Asterisk for voicemail, call routing (for long-distance, LNP, etc) and SER/rtpproxy at the other end which handles NAT onto private networks where customer's exist. This setup works fairly well for the most part, except that Asterisk does not have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy) for their jitter buffer on both ends of our voice links here. To me, that would mean shoving a SER/rtpproxy combo between Asterisk and our provider network. Possibly on the same server. I could easily throw up a SER installation, but I'm trying to figure out if there's any way to leave Asterisk in the SIP path but remove it from the media path (have RTP just go straight from SER/rtpproxy to my provider's RTP proxy). Have any of you set up a scenario somewhat like this? Any recommendations? Asterisk CVS-HEAD does let you apply a patch and get some very alpha jitter buffer support. But CVS HEAD doesn't work reliably for me at all currently so I'm sticking with the latest stable release of 1.0.x series. I'm running SER 0.9.3 FYI on the SER proxies I have set up currently. Thanks. -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

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[Serusers] Please advise...
by Stanly Levithan 30 Oct '05

30 Oct '05

Dear SER-users, We are looking to establish an educational business, namely a virtual learning center,using a conference service providing capabilities with the following calls flow: --Caller dial a local number -- Media gateway/media server (MG/MS)accepts the call and automatically mutes it so all the callers become passive listeners if number is registered in the Data Base; if the number is not registered the call is discarded --MG/MS plays Welcome message --MG/MS asks for a virtual conference room number --Caller enters virtual conference room number --Caller is entered into virtual conference room associated with the virtual conference room number. All callers entered virtual conference room number are now in associated virtual conference room --MG/MS accepts any callers decision to switch to a different virtual conference room by entering appropriate virtual conference room number without making another call (without leaving the system) --MG/MS permits the caller to terminate the call at will --MG/MS of our dream has to be able to process up to 50, 000 callers simultaneously --MG/MS provides billing. Kindly let us know if SER is capable of supporting such a calls flow. Please let us know if you need any additional information in order to advise us on a most efficient configuration. Regards, Stan *** This email was scanned by eSafe Content Inspection Server. *** __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

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[Serusers] Please advise...
by Stanly Levithan 30 Oct '05

30 Oct '05

Dear SER-users, We are looking to establish an educational business, namely a virtual learning center, using a conference service providing capabilities. Kindly take a look at our business requirements described in the attachment and let us know if SER is capable of supporting such a calls flow. Please let us know if you need any additional information in order to advise us on a most efficient configuration. Regards, Stan *** This email was scanned by eSafe Content Inspection Server. *** __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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[Serusers] SER installation procedure for Redhat linux 9.0: Please note this
by Dhiman D Chowdhury 30 Oct '05

30 Oct '05

Dear friends, If you are installing SER for the first time in REDHAT LINUX 9.0, please do the followings: 1. download ser-0.9.3_linux_i386_tar.gz 2. save it to a user directory 3. login as root and go to / directory 4.from "/" as root run the following command: tar -zxvf /home/user/ser-0.9.3_linux_i386_tar.gz [replace the /home/user directory with the directory name where you have initially downloaded the ser file. Please note ser-0.9.4_linux_i386.tar.gz will not work if you have downloaded it. For ser 0.9.4, you need to make binary by yourself. 5. Now run /usr/local/sbin/ser 6. Your SER server is running. 7. At this point you can download xten pro sofphone client and configure it to connect to your serv server. 8. Just to check out whether or not xten is registering with the server use admin as user and heslo as password. Now you are ready to connect mySQL server and follow the procedure for that. Thanks Dhiman Deb Chowdhury

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[Users] Making OpenSER give up being man-in-the-middle?
by Juan Carlos Castro y Castro 30 Oct '05

30 Oct '05

Hello, I'm using OpenSER to make load balancing among a bunch of Asterisk boxes. I whipped up a very simple script that works: # module stuff goes before here route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; route(1); break; } route[1] { exec_dset("/usr/local/bin/selectserver.sh"); if (!t_relay()) { sl_reply_error(); }; } selectserver.sh alters the server in the destination URI to choose from a dynamic pool of servers that's always changing according to a funky proprietary logic of ours. This works just peachy. The problem is OpenSER stays as man-in-the-middle all the time. I'd like it to do a REFER to a random server of our choosing and stay out of the loop from there on. How could I do that?

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[Serusers] Ser v0.9.4 error message " /lib/tls/libc.so.6: version `GLIBC_2.3.4' not found (required by ser)
by Dhiman D Chowdhury 30 Oct '05

30 Oct '05

Hi, I am using redhat linux 9.0 and installed ser v0.9.4. While I am trying to start ser, I get the following message: /lib/tls/libc.so.6: version `GLIBC_2.3.4' not found (required by ser) I will appricate your kind help in this regard. Thanks Dhiman Chowdhury

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[Users] usrloc-cl with openser
by Daryl Sanders 30 Oct '05

30 Oct '05

Greetings Everyone, Has anyone tried the Andreas' cacheless usrloc module with openser. Will it run on openser? I am thinking of testing the new openser release and I like to have this functionality if possible. Thanks! - Daryl

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[Serusers] Information on how to submit code to experimental
by Greger V. Teigre 30 Oct '05

30 Oct '05

To those of you who have developed code you are evaluating whether to commit to cvs, here is information found in the new HACKING file found in the experimental cvs directory. If you have any immediate questions or comments, I'm all ears! g-) HACKING SER =========== This information is for people who have or would like to develop a module or code for SER and would like to submit the code to the SER CVS tree. Please first read the README file in the experimental directory for information about the purpose of the directory and what type of code you can submit. If you have decided to go ahead, the information below is what you need to get started! Administrator ============= The administrator of the experimental directories is Greger V. Teigre. He can be contacted at greger(a)onsip.org. If you have any questions after reading this or would like to submit code, please send an email! Registration ============ As a new SER developer you need to register an account at http://bugs.sip-router.org/ and at http://developer.berlios.de/account/register.php. Use the same username for both. Requesting Access ================= In order to get write access to the experimental CVS you need to send information about the module/code to the administrator (see above), as well as your full name, and the username and email address used for the above registration. Information about parameters and function calls, the version of SER the module has been developed for, and one or more use cased for the code will be of help (unless it's obvious...). You should also include a 3-5 lines description of the code/module that you would like to include in the experimental README. You will then be advised on how to make sure your code is a use for others and if others are working on/requested submission of similar code. The purpose of this check is to make submitted code as easily accessible as possible and avoid overlapping coding efforts. Once approved, you check out using: cvs co experimental. You will get all the experimental modules, your module should have an empty directory where you have write CVS access. Development Requirements ------------------------ You should read this general info on the SER CVS: http://www.iptel.org/ser/cvs/ And to understand how SER is organized, you should read the first part of: http://www.iptel.org/~janakj/ser_cvs.xhtml We strive to keep the requirements as few as possible (we don't like overhead more than you do), but there are some rules you must follow: - You are responsible for maintaining the module for any branches you have submitted the module in. This includes keeping it updated as HEAD and the branches are updated - Bug and issue tracking MUST be kept on http://bugs.sip-router.org/ - If a module is not being maintained, it will be moved to the "attic" and eventually deleted - All experimental modules should be checked in with paths as if they live in the main modules directory - All experimental modules should have an updated README - All experimental modules should have #warning directives that will be shown when making, as well as a LOG(L_ALERT, "WARNING! ..."); in the initialization function of the module. This warning should also be added to the README file. Recommended warning template: "This module is experimental and may crash SER or create unexpected results. You use the module at your own risk. Please submit bugs at http://bugs.sip-router.org/" - IF YOUR MODULE REQUIRES PATCHING OF CORE OR OTHER MODULES: If the module requires a patch to the core tree, the README file and compile warning should explicitly warn about this and describe how to go about patching the tree (reference to install.sh below and README). Patch(es) should be included in a separate sub-directory called patches/. The README should have instructions on which changes will be made, and the maintainer should keep an updated install.sh that will apply the patches in the patches/ directory (assuming that the module is located in the ser modules directory after checkout by the user). This script may include patching code in root/other modules, as well as making a new directory in root and copying files from the modules/modulename/patches/ directory to the root. IF the contributed code is NOT a module, but a core extension/patch, the script should make the proper tree modifications, should NOT include a Makefile (i.e. a module makefile), and should leave its own directory intact after patching. - If and when the module has proven its usefulness to a broad range of users and its stability (and the developer his/her ability to support and maintain the module), it may be promoted to a standard module status. This is a decision made by the core developers. An application can be sent to serdev(a)lists.iptel.org if and when the maintainer finds the module applicable to such status

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[Serusers] Information on the experimental directory
by Greger V. Teigre 30 Oct '05

30 Oct '05

FYI here is the newly updated README from the experimental directory. The first section explains the purpose of the directory. g-) The Purpose of The Experimental Directory ========================================= The experimental directory has been created to lower the threshold for adding new code to SER, while keeping the main code tree robust. Experimental exists both in CVS head (the main development branch), as well as in release branches (ex. rel_0_9_0). In the CVS head, experimental can be used for patches to the main code tree, as well as for modules that extend SER's functionality. Any code will be excepted, provided that the code give some generic value and one or more developers are willing to maintain the code. Code that proves itself popular and stable will be moved to the main code tree. In a release branch, experimental can be used to add new functionality to an already released version. Code will never be allowed into the main code tree (because the code has been freezed), but people can add experimental features to the stable release. This way limited functionality can be added without having to use the CVS head, which will have lots of (possibly untested) functionality added. Often modules that have been developed for CVS head will be "backported", i.e. made to work with a previous, stable version. Also, some developers may use a stable release and have developed a module for that version. This module can be submitted to experimental of a released version to facilitate the introduction of the new code into SER. (NOTE that ALL code submitted to a stable version MUST be ported to also work with CVS head. Allowing a module only into the experimental directory of a stable release is just time-limited until porting has been done.) What You Will Find in Experimental ================================== The experimental directory may contain three types of code: 1. *Simple modules* that can be dropped into main code tree 2. *Complex modules* that require patching of the main tree 3. *Code patches* (no module) that modify the core to accomplish new functionality Submitting Bug Reports ====================== For experimental code to enter the main code tree, it needs to be used, tested, and bugs/feature requests most be reported. If you use code from the experimental directory,remember that the code maintainer will love to hear about your experiences! Contact the code maintainer directly (see information below). If you have found a bug or have a feature request, please use: http://bugs.sip-router.org/ ============================= Description of Modules: ============================= path module, simple module ----------- COMMENT:The path module has been ported into the main code tree by Andreas Granig. This module will thus be removed as this effort has been finished. Maintainer: Fermin Galan Marquez <fermin.galan(a)agora-2000.com> This module implements the Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts, as described in RFC 3327 tls, code patches ---------- Maintainer: Cesc Santasusana <cesc.santa(a)gmail.com> TLS is an optional part of the core, not a module. TLS, as defined in SIP RFC, is a mandatory feature for proxies and can be used to secure the SIP signalling on a hop-by-hop basis (not end-to-end). TLS works on top of TCP (DTLS, or TLS over UDP is already defined by IETF and may become available in the future). usrloc-cl module ----------- COMMENT: This module is currently only found in rel_0_9_0 branch. Work is in progress to port the module to CVS head. Maintainer: Andreas Granig <agranig(a)linguin.org> This is a cacheless implementation of the usrloc API and replaces the original usrloc module. It provides access to domain tables (like location and aliases) to other modules. The module exports no functions that could be used directly from scripts. The typical use case for this module over the original usrloc module is that one wants to replicate the tables on the database layer without the need of SIP replication. This allows better scalability at the cost of performance. osp module ---------- COMMENT: The maintainer is currently working on packaging the code for submission to the cvs. Maintainer: Dmitry Isakbayev <isakdim(a)gmail.com> This module adds support for the Open Settlement Protocol to SER. The Open Settlement Protocol (OSP) standard is an operations and billing support system protocol for IP network applications such as VoIP, video, short message services (SMS) and content brokering. OSP is an open standard defined by ETSI - the European Telecommunications Standards Institute. OSP has been widely deployed by VoIP carriers to enforce secure access control for peer to peer inter-domain VoIP routing and Call Detail Record (CDR) collection. For more information see www.etsi.org and search for ETSI TS 101 321. lcr module ---------- COMMENT: The LCR module of head has a slightly different feature set that have been difficult to backport to 0.9.x. Maintainer: Juha Heinanen <jh(a)tutpro.com> Backported Least Cost Routing module from HEAD.

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[Serusers] MySQL support faild - Ser can'r connect to Database - Why?
by Dirk Schäfer 29 Oct '05

29 Oct '05

Hello Friends, I have installed Ser with MySQl support and Serweb.Wenn I start the Ser in normal mode - all things are ok and the Ser don't make any failure. Whenn I turned the fork mode on , and start the Ser a second time, it show me a connection error with mysql. Ser can't connect to Database ,so that i can not register any Users with Serweb or so . But serctl work's fine. The database name and password was ser and heslo. I'm using Ser Verison 08-09 but ist still the same failure and i doesn't now, how i can make it better. I'm using the standard ser.cfg file with the normaly changes of IP-Adresse and db_mode. I hope anyone can help me with this problem. I only want to setup a Ser with Mysql Support and Serweb. Thanks and greetings, Dirk

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[Serusers] how to uninstall ser 0.9.3
by julien bos 29 Oct '05

29 Oct '05

Hi all, Can you tell me how to uninstall ser0.9.3? I want to install ser 0.9.4. i try to install ser 0.9.4, after that ser -V reponds version 0.9.3? Thank you so much for your instruction

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[Serusers] ser-rtpproxy bridge mode and T38
by Atif Rasheed 29 Oct '05

29 Oct '05

Hi everyone, I have raised this issue in dev list but got no response, hoping some one here will be able to answer this. I have read in RtpProxy that RtpProxy waits for both parties to send atleast one udp packet and then it fills up both ip:port structures associated with that call and then it starts relaying. Call Setup is like this: --------Media-----> A-Party ----->MG-------->SSW------>SER/RtpProxy------->B-Party now only B-Party is sending UDP Packets and Ser/RtpProxy sends them to SSW (signalling entity) instead of MediaGateway. As soon as RtpProxy receives one udp packet from A-Party then it sends those RTP packets to MG i.e. it starts relaying following the right path. how I can solve this ??? Adding to this, now after RTP relay starts successfully. B-Party (if Dialed number is a FAX) reinvites with T38 and the IP:Port info in SDP remains same. Now somehow RtpProxy is not recognizing the T38 packets becuase we are in the same session and RtpProxy start sending those T38 packets to SSW (signalling entity) instead of sending to MG. Please remember we are in the same session, and RtpProxy shud send these T38 packets to MG, as it was sending RTP packets before ReInvite. is it a bug in RTPProxy, should I use MediaProxy instead ??? Please comment, your help will be highly appriciated. I am using CVS version of SER and RTPProxy. Thank you Atif Rasheed

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[Serusers] pthread create failed with code 11
by Nicky 29 Oct '05

29 Oct '05

Hi Support, I have experienced this error, please advice what can be the caused. Thanks. ERROR: start (AmThread.cpp:85): pthread create failed with code 11 Regards, Nicky

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[Users] package submission
by Daniel-Constantin Mierla 28 Oct '05

28 Oct '05

Hello to everybody that volunteered to build packages, you can start creating the packages for the distributions you subscribed. Please take the tarball with the sources from: http://openser.org/pub/openser/1.0.0/src/ Inside, in the 'packaging' directory you will find the specs for several distributions. It is very likely that you have to change them a bit (see my previous post http://openser.org/pipermail/devel/2005-October/000872.html) and discover some missing items in the specs. For tls packages you have to set TLS=1 in the environment. To submit them, would be good if you have a site from where we can downloaded via http or ftp, otherwise let me know and I will give you further details. If you have some questions, send an email. Cheers, Daniel

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[Users] Insert name into from header of SIP messages
by Brandon Price 28 Oct '05

28 Oct '05

We are storing cnam (callerid name) values in a mysql database for use with openser. I have openser pulling the values from the database using avpops but I need to know how to insert the value into the from so that it will be passed to my sip user agents. Related code posed below modparam ("avpops","db_scheme","cnam_scheme:username_col=phonenumber;table=cnam_c ache;value_col=cnam;value_type=string") modparam("avpops", "avp_aliases", "cnam=s:cnam") avp_db_load("$from/cnam_scheme","cnam/$cnam_scheme"); #sample query -- select cnam from cnam_cache where phonenumber='13143212222' this should now store that value in $cnam. now how do I put that value for example "John Smith" in the from portion of sip messaging so that it displays on the users sip phone, or phone attached to an ata?

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[Serusers] How to search the serusers archives
by Paul Antinori (pantinor) 28 Oct '05

28 Oct '05

I am looking for the search feature on the serusers archives, is this available? Also, is the SER 0.9.4 red hat RPM available for download as I cannot find it on iptel.org or the onsip website. I see the tar binaries itself, but not the RPM. Thank you for help, Paul

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[Serusers] Branch structure
by Ing. Francisco Talavera 28 Oct '05

28 Oct '05

Hi, I have a question. Can I put two URI in the same Invite message?.... My Idea is this I have a module (MY MODULE), and an external program (MY_PROGRAM) those are connected by Unix socket When SER receives an Invite, Mymodule send this to Myprogram, and Myprogram rewrite de uri of the invite and send to Mymodule.. then I want to implement Forwarding. That is A(pedro) --------------------call ---------------------------> B(alice house) If B dont answer, i must forward to C(alice office).. and so on It could arrive doing 2 Uris in the same invite or I have a branch structure in the msg message? Myprogam do Accounting, routing and others Some Ideas.. Thank in advance, Ing. Francisco Talavera Dpto. Investigación y Desarrollo Conexión S.A.

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[Users] TLS changes
by Klaus Darilion 28 Oct '05

28 Oct '05

Hi! I've changed the default value of tls_disable to 1 Thus allows using the standard openser.cfg with TLS enabled openser. klaus

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[Users] debian packaging question
by Klaus Darilion 28 Oct '05

28 Oct '05

Hi! On my box there is installed: libmysqlclient10 3.23.56-3 LGPL-licensed libmysqlclient10-dev 3.23.56-3 LGPL-licensed libmysqlclient12 4.0.24-10 mysql database libmysqlclient14 4.1.11a-4sarge1 mysql database and: /usr/lib/libmysqlclient.so -> libmysqlclient.so.10 Thus, I suggest the mysql module is linked with libmysqlclient10. I further guess, others have to use the same libmysqlclient version? Is there a recommende version of libmysqlclient to use ? regards klaus

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[Serusers] bridging SER peering other Sip Servers
by Joao Pereira 28 Oct '05

28 Oct '05

Hello to all, I have my SER in bridging mode (with two IPs in two non routable networks), with rtpproxy and nathelper: rtpproxy -l 10.0.0.135/193.136.2.205 And the clients in the private and public networks can speak each other with no problems. But in the calls from the private network to external SERs (for example: user(a)iptel.org), the speech only passes one way (from my private IPclient to the user(a)iptel.org) Can someone tell me what could be the problem? These are my routing lines: if (lookup("interno")) { if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } if (dst_ip == 193.136.2.205){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } } else if (lookup("externo")) { if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } if (dst_ip == 193.136.2.205){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } } else { sl_send_reply("403", "Utilizador nao Encontrado"); break; }; (....) Thats my external routing section: if( !uri=~"@fccn.pt"){ if (!method=="REGISTER") # internal IPs must use RTP proxy to call user(a)iptel.org if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); route(1); break; } record_route(); t_on_reply("3"); t_relay(); break; } (....) onreply_route[1] { if (!(status=~"183" || status=~"200")) break; force_rtp_proxy("FA"); }

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[Serusers] how can i avoid multiple logins?
by Ryan Pagquil 28 Oct '05

28 Oct '05

Hi, How can I avoid multiple logins using the same username? I want to secure the users so that only the authentic user can only log on? Is there any modules or commands to make this? For example user1 is logged I don't want other people to log also using user1 since user1 is currently logged on. Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Users] steps to build rpm packages
by Daniel-Constantin Mierla 28 Oct '05

28 Oct '05

Hello, I try to summarize basic steps for generating the RPMs (exemplified for RedHat). The RPM experts can update/correct these guidelines. a) an archive with the sources will be available for download (openser-1.0.0_src.tar.gz) once everything is prepared b) download it in /usr/src/redhat/SOURCES c) copy the spec file that is designed for your distribution in same folder (for RedHats is better to try first openser.spec.fedora) as openser.spec d) edit openser.spec as follows: ** Release: must be set to a string that identifies the distro (e.g., for fedora core 3.0, it should be "Release: fc3.0") ** update the packager name and address to yours ** comment the packages that you cannot generate (e.g., jabber, radius) - there are several sections for each package, which can be easily identified by the name of the package -- you have to comment the whole section e) copy the init file (either openser.init or openser.init.SuSE, from sip-server/packaging/rpm) in /usr/src/redhat/SOURCES f) cd /usr/src/redhat/SOURCES g) rpmbuild -ba openser.spec For SuSE there might be some small differences, but basically is the same process. These guidelines are for those to have root access. If not, then you have to follow first http://dag.wieers.com/howto/bits/rpm-build-user.php (more help via google searching after "building rpm as user"). If everything goes fine, you should get at the end something like (e.g., I built openser base and mysql package): Wrote: /usr/src/redhat/SRPMS/openser-1.0.0-rhel3.0.src.rpm Wrote: /usr/src/redhat/RPMS/i386/openser-1.0.0-rhel3.0.i386.rpm Wrote: /usr/src/redhat/RPMS/i386/openser-mysql-1.0.0-rhel3.0.i386.rpm These are the files we need. If possible, please make them available on a ftp or http site and send me the address to take them. I will notify you back onces I got them, to remove them from your location, if you wish. Otherwise, I will accept attachments to email (at this address), but let me know first that you are going to send them, to be prepared :-) Cheers, Daniel

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[Serusers] How does user agents send log-off request?
by Ryan Pagquil 28 Oct '05

28 Oct '05

Hi, How does User agents sends log-off signal to SER? I know that for the user agent to log in it needs to send a REGISTER request, what request should the user agent need to send SER to be able to log off and be removed from the location table? Could you please explain how the user agent does this... Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] question on Adding aliases
by Eric Haskins 28 Oct '05

28 Oct '05

We are trying to get inbound DID's working on our SER box. Outbound is working perfectly as we are doing pre-paid. I ran this command [root@voip1 ~]# serctl alias add 6049094251 sip:5002@216.x.xxx.xx domain unknown: use usernames with domain or set default domain in SIP_DOMAIN [root@voip1 ~]# I get the same result if I put our domain in place of the IP address. We had someone else set this box up for us and I am having problems reaching them. 5002 is is the username of my test ATA logged into the system I can make outgoing calls to the PSTN What am I doing wrong?? I have been reading Archives allday trying to grasp this. The DID provider is directing that DID at the IP of this SER box Eric Haskins Lead Software Developer eric(a)v1010.com

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RE: [Serusers] end the session if no media
by Hakan YASTI 28 Oct '05

28 Oct '05

Yes, I know about it. I can see the active sessions from the session.py. But do not know how to end the seesion, if there is no rtp session and the bye has not come. My session con troller does not check the rtp session because the media is proxied at SER. As I know, may be I am wrong, the ser is managing the media proxy. Ser says to media proxy to start when a call is proxied, and end when the call is finished. Do you suggest a SBC that can do this function..? Thanks, Best Regards, ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, October 27, 2005 9:20 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media session.py will show if the session is active. Also when a call is put on hold, there will be no rtp stream. I don't know if your SBC handles that. Or is your question something else? g-) ----- Original Message ----- From: Hakan YASTI <mailto:hakan.yasti@teletek.net> To: Greger V. Teigre <mailto:greger@teigre.com> ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Wednesday, October 26, 2005 09:18 AM Subject: RE: [Serusers] end the session if no media Hi Teigre, Thanks for your reply and kind suggestions. In fact, I am using a session controller for sip calls. And it alerts me via email when a sessiıon is reached for about 60 minutes. At that time, I check it manually and kil the session if the session is abondened. By sending BYE messages to both legs. From the PSTN side I get the reply what I assume ( 200 OK ) but from the sip side I get No transaction exist. Because the session was ended before and session controller couldn’t get the BYE message for some reasons. ( Network failure or something else ). What I am looking for that, from the media proxy, I see the active rtp session when I use media proxy for the calls. And I assume that, when the call has finished and couldn’t send or get BYE message, there will be no rtp stream going on. May be I am wrong for this. If rtp session is still open when the a side hunp up and couldn’t send bye , there is nothing to do except killing the session with sending bye messages to both sides, because there is no way to understand the call is active or not. Best Regards, Hakan ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, October 26, 2005 10:00 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media Hakan and Ritesh, It seems to me that you are both asking about the same thing. This topic has been discussed many times on serusers. Search google (site:mail.iptel.org) for "BYE stateful sipsak", I'm pretty sure you will get some info. Also, this FAQ could help: http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 g-) ----- Original Message ----- From: Hakan YASTI <mailto:hyasti@teletek.net> To: serusers(a)lists.iptel.org Sent: Thursday, October 13, 2005 09:13 AM Subject: [Serusers] end the session if no media Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net <mailto:hakan.yasti@teletek.net> http://www.teletek.net ________________________________ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir.

2 1

[Serusers] Beer on Wednesday 26th
by Atle Samuelsen 28 Oct '05

28 Oct '05

Hi Guys And Girls, Fellow Serusers, Greger and Me where thinking of trying to join the norwegian troops at Beer Palace on Aker Brygge at 5pm, Upcoming Wednesday. Anybody Intrested in Joining for a Beer or 5? Best Regards Atle

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[Serusers] Removal of email
by voip learner 28 Oct '05

28 Oct '05

Dear sir, Please remove me from the list of the emails. Iam getting hundredsof emails and the mail folder is getting bulk. Kindly do it ASAP. Regards __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] Re: Open source billing solution for SER (VoIP)
by Eric Haskins 27 Oct '05

27 Oct '05

>Hello experts, > > I was trying to set up a VoIP system with SER as my signalling >proxy and was able to get lot of useful info from this mailing list. >Thank you. > I would like to know if there exists an open source billing >solution which can fit seemlessly with SER and can handle realtime >rating ( in a prepaid scenario). My initial study on this gave me the >following link with a list of open source billing solutions. I would >also like to know some reivews on these from experts like you. Since I >have very less time to bring my project to life. > >http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems > > >Thanks a lot in advance. > >Regards, >Jithu We have a working system we built that allows for rates to be preset in a db. It calculates start and stop using acc table. I dont know if we will release as gpl but if you want to see the backend functionality contact me directly. It is already setup to work in prepaid eric(a)v1010.com Eric

1 0

[Users] TCP-alias and NAT - one more try to get an answer
by Alexander Ph. Lintenhofer 27 Oct '05

27 Oct '05

Hi All, I just wanted to ask you once again about the TCP-alias riddle. I found out, that there is a problem with the combination of fix_nated_contact(), force_tcp_alias() and NAT: Imagine following situation: Alice behind NAT: socket 172.16.0.6:2421 Nat-Box translates this to 192.168.0.13:6007 The Outbound-Proxy of Alice is 192.168.0.1 Bob is registered with 192.168.1.1 1.) Her INVITE: ==================================================================== INVITE sip:bob@192.168.1.1:2331;transport=tcp;line=wxqurd1s SIP/2.0 [...] Via: SIP/2.0/TCP 172.16.0.6:2421;received=192.168.0.13; branch=z9hG4bK-wm9jcstcboys;rport=6007 From: "Alice" <sip:alice@atlanta.com>;tag=sufzmxi0us To: "Bob" <sip:bob@biloxi.com>;tag=c9550czwtn [...] Contact: <sip:alice@172.16.0.6:2421;transport=tcp;line=fyyuh6tl> [...] ==================================================================== 2.) fix_nated_contact() doesn't work with TCP (look at nathelper.c). force_tcp_alias() now creates following tuple as TCP-alias: 192.168.0.13:6007 to 192.168.0.13:2421 Reason: The TCP-alias is not built solely from the Via-header as suggested in the draft. The portnumber is taken from the Via-header and the IP-address is taken from the source of the incoming datagram. I read it in the sourcecode and assured it by contacting Andrei! 3.) So as a result of the notfixed Contact-header of Alice's INVITE the BYE of Bob is addressed to 172.16.0.6:2421. But no TCP-alias exists for this socket :-( 4.) I made following test by rewriting the Contact-header.... ==================================================================== if (method=="INVITE") { replace("Contact: <sip:alice@172.16.0.6:2421;transport=tcp;", "Contact: <sip:alice@192.168.0.13:2421;transport=tcp;"); } ==================================================================== ...with success. Now TCP-alias works as you can see on my Ethereal-trace below! Compare the destination port of the packet to the destination port of the RURI! ==================================================================== [...] Transmission Control Protocol, Src Port: 5060 (5060), Dst Port: 6007 (6007), ... Session Initiation Protocol Request-Line: BYE sip:alice@192.168.0.13:2421;transport=tcp;line=fyyuh6tl SIP/2.0 Message Header [...] From: "Bob" <sip:bob@biloxi.com>;tag=kcsveifugd To: "Alice" <sip:alice@atlanta.com>;tag=ricaq5cy15 Contact: <sip:bob@192.168.1.1:2331;transport=tcp;line=wxqurd1s> [...] ==================================================================== Another solution: Comment the lines in nathelper.c which force the return in case of TCP or TLS. Now all works well! But why?????????????? regards, Philipp

5 10

[Serusers] LDAP authentication (without Radius)
by Felipe Louback 27 Oct '05

27 Oct '05

I'm working in a VoIP solution here in my university and we have a wide LDAP base and I am thinking about using it for I found on the internet Ldap/Ldaps modules for SER (http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldaps), but I am afraid that I could only use them for lookups and not direct authentication. I guess I could use radius for authentication but I dont know if installing a radius server "just" for this would be worth the effort. Does anyone know if there is a way to authenticate users in a LDAP server without using Radius? I mean, could I use SER directly with LDAP without using RADIUS? Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

4 6

[Users] openser 1.0.0 packages generation
by Daniel-Constantin Mierla 27 Oct '05

27 Oct '05

Hello everybody, I would like to ask all of you that have access to different OS distributions to help with package generation. Please reply (personally if you prefer better this way) with OS name, version and architecture. We are looking mainly for common used distributions: Debian - testing (the other two are accounted already) Redhat Suse Open/NetBSD (FreeBSD accounted) SUN/Solaris Gentoo ... We will help as much as possible with guidelines. We would require tomorrow an early compiling on all available systems to fix eventual warnings. Cheers, Daniel

4 10

[Serusers] SER consultant wanted
by Kevin 27 Oct '05

27 Oct '05

I am looking for a consultant to help with a SER development project. - Kevin

1 0

[Serusers] are voicemail and msilo module the same?
by Meidiana Ten 27 Oct '05

27 Oct '05

Hi there, i'm new here and need help to fix my problem. I have final project that is configuring msilo module in SER 0.9.1. But i'm new in Linux and don't have any idea bout it. i want to ask if: 1. voicemail and msilo module the same? 2. how can i make the very first step? umm..begin from the installation of modules that i need to configure msilo module i'm blank...please someone help me.. thanx.. Yours, Meidiana email add : gotcha_em(a)yahoo.com Indonesia

1 0

[Serusers] IAS and SER
by Andrej Sirk 27 Oct '05

27 Oct '05

Hello, is there somebody who made an working authentication with SER and Microsoft's IAS.

1 0

Re: RE : [Serusers] OpenSER infrastructure dimensioning
by Daniel-Constantin Mierla 27 Oct '05

27 Oct '05

The mailing list has no archive, it is created specially in such way. See: http://openser.org/pipermail/users/2005-September/000941.html Daniel On 10/27/05 17:11, Olivier Taylor wrote: > The Business Archives > > No messages have been posted to this list yet, so the archives are currently > empty. You can get more information about this list. > > Olivier > > -----Message d'origine----- > De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la > part de Daniel-Constantin Mierla > Envoyé : jeudi 27 octobre 2005 16:03 > À : Vanlandschoote Michel > Cc : serusers(a)lists.iptel.org > Objet : Re: [Serusers] OpenSER infrastructure dimensioning > > > Hello, > > On 10/27/05 16:13, Vanlandschoote Michel wrote: > >> Hi, >> >> we are considering OpenSER for offering free SIP-based voice, video >> and IM services and I am trying to get an idea of the project cost. >> >> The components we need are: >> - SIP stateful proxy + registrar server >> - location & presence server >> - simple2jabber >> - media proxy for symmetrical NAT traversal >> - (perhaps I forgot something here?) >> We'll rely on external gateways for PSTN access so that is not to be >> taken in consideration here. >> >> Now can somebody give me an indication on the server park >> dimensioning? In other words, how many SIP servers, media proxies, ... >> would I need per 10000 subscribers? >> > a computer as you described below is more than necessary. The bottle > neck could be the bandwidth if you proxy the media stream through the server > >> >> The server platform could be e.g.: >> - operating system: Linux >> - hardware: HP ProLiant DL380 Generation 3 (single CPU with 1GB RAM) >> but of course any equivalent would do as well. >> >> I also have a second question: which organisations in Europe offer >> commercial support for OpenSER? >> > The openser project is the result of many contributions, its public > activity and development is guided by business independence. For such > requirements and questions, the project provides a mailing list > business(a)openser.org, where the contributors and independent consultants > are ready to take your request: > http://openser.org/cgi-bin/mailman/listinfo/business > > > Daniel > > > >> >> Hope this forum is the right place for this kind of questions... I >> really appreciate your input, >> >> Michel >> >> ************************************************************* >> >> Dit e-mail bericht inclusief eventuele ingesloten bestanden kan >> informatie bevatten die vertrouwelijk is en/of beschermd door >> intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd >> voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit >> bericht (waaronder de volledige of gedeeltelijke reproductie of >> verspreiding onder elke vorm) door andere personen dan de >> geadresseerde(n) is verboden. Indien u dit bericht per vergissing >> heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit >> bericht te verwijderen. >> >> This e-mail and any attachment thereto may contain information which >> is confidential and/or protected by intellectual property rights and >> are intended for the sole use of the addressees. Any use of the >> information contained herein (including but not limited to total or >> partial reproduction or distribution in any form) by other persons >> than the addressees is prohibited. If you have received this e-mail in >> error, please notify the sender and delete its contents. >> >> ************************************************************* >> >> ---------------------------------------------------------------------- >> -- >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] variables with avp_check
by Joao Pereira 27 Oct '05

27 Oct '05

Hello, I want to use variables in ser.cfg and I m trying to use avpops for it # first by giving the variable "var" the value "internal" # string mysar = internal; avp_write("internal", "s:myvar"); # and checking this variable later: # if (myvar = internal) if (avp_check("s:myvar", "eq/internal/i")){ But this never enters in this if, can someone tell me the way to do it? Or where is some documentation about avpops (iptel.org/ser doesnt have). Thanks Joao Pereira

1 0

[Serusers] OpenSER infrastructure dimensioning
by Vanlandschoote Michel 27 Oct '05

27 Oct '05

Hi, we are considering OpenSER for offering free SIP-based voice, video and IM services and I am trying to get an idea of the project cost. The components we need are: - SIP stateful proxy + registrar server - location & presence server - simple2jabber - media proxy for symmetrical NAT traversal - (perhaps I forgot something here?) We'll rely on external gateways for PSTN access so that is not to be taken in consideration here. Now can somebody give me an indication on the server park dimensioning? In other words, how many SIP servers, media proxies, ... would I need per 10000 subscribers? The server platform could be e.g.: - operating system: Linux - hardware: HP ProLiant DL380 Generation 3 (single CPU with 1GB RAM) but of course any equivalent would do as well. I also have a second question: which organisations in Europe offer commercial support for OpenSER? Hope this forum is the right place for this kind of questions... I really appreciate your input, Michel ************************************************************* Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. Indien u dit bericht per vergissing heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit bericht te verwijderen. This e-mail and any attachment thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the sole use of the addressees. Any use of the information contained herein (including but not limited to total or partial reproduction or distribution in any form) by other persons than the addressees is prohibited. If you have received this e-mail in error, please notify the sender and delete its contents. *************************************************************

3 2

[Serusers] Curious interop bug.
by David Gilbert 27 Oct '05

27 Oct '05

For those of you who want to follow along, the packet dump in question is at http://strike.dclg.ca/~dgilbert/22-3-filtered ... it's a pcap format file. I'm using a 0.9.4 copy of ser, compiled by hand on debian stable. Only the core portion of ser is being used, and I have simple config file (attached). So, roughly, a call comes in level3, they make it SIP and forward it to my SER on port 5070. SER turns around and forwarded it to asterisk on port 5060 of the same machine and then asterisk forwards it on to a linksys ATA. In short, the problem is that level3 abuses the RFC by sending from a port and expecting you to reply on another port (specified in the Contact: header). This goes almost right except for level3's equally literal read of the Contact header that SER sends. Packet Sip Command Contact ====== =========== ======= level3:38999 ser:5070 INVITE sip:+1416...@<level3 ip>:5060 ser:5070 level3:38999 100 Trying sip:+1866...@<ser ip> (ser invites asterisk, not in dump) asterisk:5060 ata:5060 INVITE sip:+1416...@<asterisk ip> ata:5060 asterisk:5060 100 Trying None ata:5060 asterisk:5060 180 Ringing None (asterisk relays 180 to ser, not in dump) ser:5070 level3:38999 180 Ringing sip:+1866...@<ser ip> ata:5060 asterisk:5060 200 OK sip:604...@<ata ip>:5060 asterisk:5060 ata:5060 ACK sip:+1416...@<asterisk ip> (asterisk relays 200 OK to ser, not in dump) ser:5070 level3:5060 200 OK sip:+1866...@<ser ip> level3:5060 ser:5060 !?! ACK sip:+1866...@<level3 ip>:5060 Now... the problem is that each time ser repsonds, the contact line should be appended with ":5070" and it is not. Therefore, being the literal people they are, level3 repsonds to port 5060 with their ACK. This means that the ACK goes directly to asterisk rather than to SER and it means that when the call ends, the BYE notification from asterisk goes directly to the wrong port at level3. Dave. -- ============================================================================ |David Gilbert, Independent Contractor. | Two things can only be | |Mail: dave(a)daveg.ca | equal if and only if they | |http://daveg.ca | are precisely opposite. | =========================================================GLO================

3 2

re:[Users] SIP clients and DNS SRV
by Zen Kato 27 Oct '05

27 Oct '05

Hi, If we set 'Use DNS SRV: Yes' and 5 minutes REGISTRATION interval on BT101(1.0.7.11) and GXP2000(1.0.1.9),these UACs can successfully Re-REGISTER and INVITE other UACs on time period B, which comes after ser1 stops, every UACs changed their target from first DNS SRV(sip1) to second DNS SRV(sip2). But during time period A if caller is not Re-REGISTERed on sip2, when he dials some number, there is 'no tone(white tone?)' this means the server is down. If the caller's phone is Re-REGISTERed on sip2 and callee's is not yet Re-REGISTERed on sip2 in the time period A, the caller's phone displays "404 not found". So, this is first 5 minutes out of service probabilty time. After the operator of servers room knew that sip1 went down and sip2 is take care of sip calls, he tries to repair sip1 and restarts 'ser(openser)' on sip1. But all INVITEs are handled on sip2 now. BT101s and GXP2000s never automatically go back to sip1. So, the operator must stop sip2 untill all REGISTERs move from sip2 to sip1. This is the second 5 minutes out of service probability time. So, (5+5)minutes out of service probability time exist. If we set 60 minutes(default) for REGISTRATION interval, and all Re-REGISTERs happen on 50 minutes after sip1 down, (50+50)minutes out of service probabilty becomes high. stop ser1 stop ser2 | | V V ....|<-- 5 min-->|<--sip2-->|<-- 5min -->|<--sip1--.....-> |service stop|service |service stop|service | |continue | |continue time|<----A----->|<---B---->|<----C----->|<----D---.....-> Do we need another backup method like Linux-HA if UACs(ONE outgoing line only devices) don't support DNS SRV? Regards, Zen ------------ quote from Daniel's question ----------- Daniel-Constantin Mierla daniel at voice-system.ro Wed Oct 26 16:46:32 CEST 2005 I do not get the question regarding the out of service interval when the client does not support srv. How you computed the 5+5 interval? ----------------------------------------------------

2 1

[Users] rport in via header
by ranveer kunal 27 Oct '05

27 Oct '05

hi all, Is there a way to configure openser so that it sends rport in via header ? Actually th packet from server is reverse natted through another system using any random port, i want it the reply to come on the same port . thanks .... Regards, Ranveer. -- Memories : They bring Diamonds and Rust

3 5

[Users] Detecting runaway calls
by Daryl Sanders 27 Oct '05

27 Oct '05

Other than Session-Timers or B2BUA is there any way to detect a runaway call? I use third party PSTN providers so, I do not control the PSTN gateway. I have read everything I can find on the lists but haven't had much luck. I know I can pull info from the acc table and fake a BYE to disconnect the call, but is there any way to first determine if the call is still actually in progress or if it a runaway. I have heard rumor of some providers that automatically drop calls after an hour or so. This is obviously not a good solution. As always, your help is most appreciated! - Daryl

3 2

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