sr-users July 2004

sr-users@lists.kamailio.org

178 participants
449 discussions

[Serusers] ENUM web lookup page
by AG Projects support 24 Jul '04

24 Jul '04

Axel, I tried +878102233342019 and it returned nada. Adrian > i don't know which domain those guys use, but my phone number in > the official e164.arpa domain was not found. I have a simple e164.arpa lookup script on my personal page: http://nona.net/features/enum/ comments appreciated. cheers axelm

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[Serusers] gsm6.10 support with ser?
by ser die 24 Jul '04

24 Jul '04

hello friends, is gsm 6.10 is supported in the sip express router with regards ser die hard __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail

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[Serusers] Another question regarding Asterisk & SER
by Curt Moore 24 Jul '04

24 Jul '04

As a preface, my setup is somewhat unique; if anyone has a working SER configuration for this scenario, or is willing to help come up with one, I would be willing to compensate you for your efforts. If you're interested, please contact me off list for details. At present, I'm using Asterisk to run a small VoIP network with great success. My focus is now to look at handling things such as NAT and scalability/reliability. This where I believe SER will be very beneficial as it handles NAT and scaling very well. My desired setup is as follows (the only thing behind a NAT are phones): Phone(s) --> [NAT] --> SER --> Asterisk(FS) --> Asterisk (PSTN) All phones behind the NAT have their outbound proxy set to SER and their regular proxy set to Asterisk(FS). SER then proxies outbound traffic from the phone to the Asterisk(FS) server acting as the "feature server" ie. VM, conferencing, etc. The feature server then routes traffic according to the dialplan in Asterisk. If the call is destined for the PSTN, the dialplan logic in Asterisk(FS) sends an INVITE to Asterisk(PSTN) which tells Asterisk(PSTN) to reINVITE the Phone at the public IP of the NAT and also tells the phone, via SER, to reINVITE Asterisk(PSTN) at its public IP. The media streams should then flow between the phone and Asterisk(PSTN) while both SER and Asterisk(FS) are out of the picture until the call ends and BYEs are processed. If a call is destined for another SIP device, the process is the same except that Asterisk(FS) forwards traffic back to the address of the phone, which is SER since SER is acting as the proxy for all SIP devices. SER then forwards the traffic back to the NATted phone as appropriate. Amazingly, I already have what I believe to be an SER config which correctly routes all of the SIP/SDP traffic as desired. My current issue is that during a call intitated by the NATted phone, the phone and Asterisk(PSTN) have reinvited and are sending RTP to each other but the phone behind the NAT cannot hear any of the audio being sent by Asterisk(PSTN) while the phone on the PSTN can hear audio being sent by the NATted phone; one way audio. What is really odd about this is that when the PSTN user first answers the call, approximately 1-2 seconds of audio is heard on the NATted phone from the PSTN user but then no more is heard for the remainder of the call. For me, I see this as some sort of RTP issue as the first second of voice RTP traffic is doing exactly what is expected between the phone and Asterisk(PTSN). I've verified the SRC and DST of the RTP traffic with Ethereal. I've also tried force_rtp_proxy and rtpproxy with no success. It is my understanding that my desired setup is very similar to that of others such as FWD (Free World Dialup), et al. My SER config follows. It is based on an Asterisk+SER configuration someone posted earlier so please point out anything you see which may not be correct. I'll also add that this configuration works flawlessly whenever there is only a phone and the 2 Asterisk servers; ie. no SER and no NAT. All reINVITES take place as they should and all media is heard as it should be. Thanks in advance! -Curt #alias=" mydomian.com " #Alias="192.168.10.100" #ser #Alias="192.168.10.120" #Asterisk check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=438 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/acc.so" # !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # set reporting log level modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { if(search("^Contact:(.*)192\.168(.*)")) { log(1,"We're natted\n"); # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling save_noreply("location"); if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save_noreply("location"); break; }; # lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay # record_route(); if (isflagset(6)) { fix_nated_sdp("1"); force_rtp_proxy(); t_on_reply("1"); } if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); fix_nated_sdp("1"); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) # } else if (nat_uac_test("1")) { } else if (search("Contact:.*192\.168.*")) { fix_nated_contact(); }; }

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[Serusers] uri in version 12
by Charles Galea 24 Jul '04

24 Jul '04

Hi, we encountered problems when trying to load the uri.so module using SER version 8.12. All other modules that we are using are loading without problems, including mysql.so, auth_db.so, auth.so, acc.so etc; is there anything we are missing? The db_url is set to modparam("uri", "url_db", "sql://serro:47serro11@localhost/ser") In version 8.13 we are using uri_db for the check_from() function and we encountered no problems there, but we are very curious to know what's wrong with the uri in version 12. Note that for version 12 we have used the rpm packages in the folder "latest" for Fedora. Regards and many thanks, Charles and Andrew

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[Serusers] what the is the module name of acc.so
by Linda Xiao 24 Jul '04

24 Jul '04

Hi, I am trying to enable the missed call logging feature in ser. When I simply add the following single line modparam ("acc", "db_url", "sql://ser:heslo@localhost/ser") in the existing ser.cfg, and restart SER, it returns "bad config file (1 errors)" I have not problem to use it for other modules, like group, tm, etc. By the way, the version of SER I am using is 0.8.12. I guess the module name got changed instead of "acc". Thanks and Best Regards Linda

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RE: [Serusers] what the is the module name of acc.so
by Dave Bath 24 Jul '04

24 Jul '04

Sorry, this may be obvious, but did you load the module at the beginning of the file? Usually something like loadmodule "/usr/lib/ser/modules/acc.so" Dave ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Linda Xiao Sent: 24 July 2004 01:17 To: serusers(a)lists.iptel.org Subject: [Serusers] what the is the module name of acc.so Hi, I am trying to enable the missed call logging feature in ser. When I simply add the following single line modparam ("acc", "db_url", "sql://ser:heslo@localhost/ser") in the existing ser.cfg, and restart SER, it returns "bad config file (1 errors)" I have not problem to use it for other modules, like group, tm, etc. By the way, the version of SER I am using is 0.8.12. I guess the module name got changed instead of "acc". Thanks and Best Regards Linda

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[Serusers] SEMS Voicemail Vs SER vm module
by Koyama Zehi 23 Jul '04

23 Jul '04

Dear all, I am a newbie in SIP please excuse me, for my poor questions. As I am working to add Voicemail service on my SER proxy. I am very much confuse about how to use SEMS module with SER. As according to "SIP Express Router v0.11 --Admin Guide" SEMS is required to provide voicemail capability in SIP server. As i have downloaded "sems-0.1.0.tar.gz" and installed in my machine where I have "ser-0.8.12" as a sip server. after that And I have changed my ser.cfg scripts to provide voicemail capabilities and to load vm module available in SER. Now I get stuck How I have to use SEMS module ( I mean to say that "ans_machine" available in sems ) what is the difference & similarity between SEMS Voicemail and SER vm module. On what basis SEMS Voicemail and SER communicate to each other. Is their any configuration file is required inside the SEMS like SER (ex. ser.cfg). what is the proper step to perform a testing of voice mail using these modules.. Please excuse me if i have asked something wrong and please teach me about my queries. Thanks in advance. koyama --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!

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[Serusers]Called phone continues ringing when caller hang up (newbie)
by vagabond 23 Jul '04

23 Jul '04

Hi, I'm running ser-0.8.11 on a Gentoobox. When phone1 calls phone2 and hangs up when phone2 is ringing phone2 doesn't stop ringing. If the person who haves phone2 answers before it stops ringing, the conversation continues with no problem. The two phones are Zyxel P2000W. Do you see what is the problem? Thanks, Alexandre My ser.cfg : debug=3 fork=no listen=192.168.1.36 log_stderror=yes check_via=no dns=no rev_dns=no port=5082 fifo="/tmp/ser_fifo" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (method=="REGISTER") { save("location"); break; }; if (method=="INVITE") { if (uri =~ "sip:1003@*") { forward(192.168.1.35, 5060); break; }; if (uri =~ "sip:1004@*") { forward(192.168.1.37, 5060); break; }; }; }

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[Serusers] Re: Hiding my number when
by Adrian Georgescu 23 Jul '04

23 Jul '04

If this is meant for calls to/from PSTN this it could be possible by toying with the privacy and screening indicators from Remote-Party-Id. http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/ 122newft/122t/122t13/ftsipext.htm#wp1027186 Adrian >>>>>>>>>>>> ok but using SER, is it really the only way to hide one's number when calling? no one else has succeded doing this maybe using some prefix like '#31#' or so? i can't believe NOBODY uses this kind of feature.. Enrique- Juha Heinanen <jh at tutpro.com> escribió: > ser writes: > > > What developpement would be needed to implement the possibility for > a > > caller to hide his number when calling via ser ? > > if you would like to modify the from field, you would need a b2bua. > you > can also try if using Anonymous from field would work in your > environment.

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[Serusers] Re: Using Asterisk as mediagateway with or without
by AG Projects support 23 Jul '04

23 Jul '04

Leon, 1. Media is normally not accepted by the gateway if is not already negotiated in the signalling. So nobody would be able to get in the middle unless they have control of the signaling which you took provision to protect already. So you are on the safe side unless there are bugs in your gateway. 2. If you use a media session controller you can enforce more checks in there and allow media ports at PSTN gateway originating only from that session controller same as you did for Mvg, Adrian >>> Does anyone have an answer to this ? It's not really SER or Asterisk related, but more generic about security for a mediagateway.. Regards, Leon On Tue, 2004-07-20 at 10:43, Leon de Rooij wrote: > Hi again :) > > Got one more question about using a mediagateway. Right now I've got > everything configured that SER relays the call to our mediagateway > (asterisk) when necessary. The mediagateway is also on a public IP, but > only accepting UDP port 5060 connections from the SER proxy. (We use RP > (reverse path) filtering on our routers, so the IP address cannot be > spoofed). Come to think of it, I can additionally also filter on MAC > address since both machines are in the same LAN.. > I read that a lot of people use an RTP proxy for forwarding the RTP > traffic to the gateway (which in turn is in a private net). > > My question is: Is my setup less secure than using the RTP proxy ? If > so, why ? > > Thanks ! > > Regards, > > Leon

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sr-users July 2004