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37529 discussions

Re: [Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 24 Feb '07

24 Feb '07

i had : UA1 (192.168.4.66:5061) }------------->Asterisk----------->SER-----------> UA3 (192.168.4.65:5061) UA2 (192.168.4.66:5062) then UA1 moved to SER domain : UA3 (192.168.4.65:5061) UA2 (192.168.4.66:5062) ------------->Asterisk----------->SER-----------> UA1 (192.168.4.66:5061) UA1 registers to SER and SER forwards the registration to ASTERISK, Asterisk is aware of the new location. UA2 tries to call UA3. It sends the invite to Asterisk and Asterisk replies with a TRYING message. after a while the call fails with an error 2007/2/24, ram <talk2ram(a)gmail.com>: > > then register with SER, should do the job > > UA----ASterisk----SER-----Phone > > is this correct > > ram > > > On 2/24/07, karim basraoui <basraouik(a)gmail.com> wrote: > > > > Hi, > > > > the call i'm trying to establish is from ASTERISK to SER. > > the UA i'm calling was initially in ASTERISK domain and moved to SER > > domain , a roaming case. > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > Hi > > > > > > your question is not clear > > > > > > how is the call path happening from ASterisk or from SER > > > or both are gateway proxies ? > > > > > > ram > > > > > > > > > On 2/23/07, karim basraoui < basraouik(a)gmail.com > wrote: > > > > > > > Hi, > > > > > > > > I've set up a small architecture in which an ASterisk proxy is > > > > connected to a SER proxy in two different domains. > > > > > > > > I have sip phones connected to both of them. when an asterisk > > > > subscriber "travels" to SER domain it tries to register to SER. I've > > > > configured SER to accept foreign > > > > subscribers and to REGISTER them to their initial domain by the > > > > command "t_replicate". Till now everything works good, the UA is registered > > > > and Asterisk is aware > > > > of its new location, it displays in peers list the UA and its new IP > > > > address and port. > > > > > > > > The problem is when i'm trying to call the foreign UA from an > > > > Asterisk subscriber, it doesn't work. i have a TRYING response but no call > > > > establishement. > > > > > > > > how can i overcome this problem please? > > > > > > > > -- > > > > BASRAOUI Karim > > > > www.basraouik.afrikart.net > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > > > -- BASRAOUI Karim www.basraouik.afrikart.net

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Re: [Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 24 Feb '07

24 Feb '07

Hi, the call i'm trying to establish is from ASTERISK to SER. the UA i'm calling was initially in ASTERISK domain and moved to SER domain , a roaming case. 2007/2/24, ram <talk2ram(a)gmail.com>: > > Hi > > your question is not clear > > how is the call path happening from ASterisk or from SER > or both are gateway proxies ? > > ram > > > On 2/23/07, karim basraoui <basraouik(a)gmail.com> wrote: > > > Hi, > > > > I've set up a small architecture in which an ASterisk proxy is connected > > to a SER proxy in two different domains. > > > > I have sip phones connected to both of them. when an asterisk subscriber > > "travels" to SER domain it tries to register to SER. I've configured SER to > > accept foreign > > subscribers and to REGISTER them to their initial domain by the command > > "t_replicate". Till now everything works good, the UA is registered and > > Asterisk is aware > > of its new location, it displays in peers list the UA and its new IP > > address and port. > > > > The problem is when i'm trying to call the foreign UA from an Asterisk > > subscriber, it doesn't work. i have a TRYING response but no call > > establishement. > > > > how can i overcome this problem please? > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- BASRAOUI Karim www.basraouik.afrikart.net

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[Serusers] iptel.org ANNOUNCEMENT: New SER bug marshal
by Greger V. Teigre 24 Feb '07

24 Feb '07

Hi all! I'm pleased to announce that Lakmal Silva has volunteered to be our SER bug marshal! Lakmal's main task will be to interface with the community and developers and make sure that bug fixes and feature requests don't disappear on the serusers and serdev mailing lists, as well as get them properly described/documented. (Please continue to use the tracker or mailing lists as you are used to.) The bug marshal's responsibilities are in more details: 1. Scan the mailings lists for bug reports, patches, feature requests, and improvement suggestions and: a) make sure they are registered on either http://iptel.org/ser/wishlist or as a tracker item (unless a developer picks it up) with unscheduled status b) request more information/debug traces/patches/etc in order to make the item possible to act upon for a developer c) solicit input from the developer(s) if clarification is needed 2. Pick up discussions on the mailing lists that concern new features, put new features on the wish list and if appropriate create a new specification/discussion page that will specify the feature in more detail and can be used for discussion purposes (examples: http://www.iptel.org/ser/specifications) Please don't contact Lakmal in private emails, but use the mailing lists if you have questions/issues. On behalf of the SER community, I welcome Lakmal and wish him luck in his new role as SER bug marshal! Greger

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[Serusers] SER 0.9.6 with H.264 Problem!
by Steven Wong 24 Feb '07

24 Feb '07

Hello All: You will find enclosed our ser.cfg file that I have adapted from a very generous SIP user from the Internet. This ser.cfg file work perfectly with Audio (IP Phone and Softphone). But when I am trying to use a Video Phone (we did try with GrandStream GXV-3000 Video Phone), it just giving us Audio but NOT Video. I did also try the same GrandStream GXV-3000 Video Phone with FreeWorldDialup and it is working fine with Video. We have checked the etherreal output while trying to make a call and we are having this type of errors on the etherreal: ----- RTP Protocol Source IP (The IP of the Video Phone) and Destination IP (The IP of the SER Server) ------ Payload Type = Unknown (126) ICMP Protocol Source IP (The IP of the SER Server) and Destination IP (The IP of the Video Phone) ------Destination Unreachable (Port Unreacheable) These 2 errors repeated six time before it stop with a good signal of: Payload Type = ITU-T G.711 PCMU Then the audio start but without audio. --- Would you please tell us what is going wrong with this ser.cfg file? your answer is greatly appreciated. Many thanks, Steven Wong -------------------------------------------------------------------------- debug=3 fork=yes log_stderror=no check_via=no dns=no rev_dns=no listen=XXX.XXX.XXX.XXX # INSERT YOUR IP ADDRESS HERE # port=5060 children=3 alias="HOST.DOMAINNAME.COM" alias="XXX.XXX.XXX.XXX" fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:PASSWORD@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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:PASSWORD@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #---Accounting params modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------------------- 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")) { # 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 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 # use different transport protocol 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; }; setflag(1); 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("HOST.DOMAINNAME.COM", "subscriber")) { www_challenge("HOST.DOMAINNAME.COM", "0"); break; }; save("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 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(); 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")) { fix_nated_contact(); }; } -------------------------------------------------------------------------- --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.

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[Serusers] How to start 2 SER at once?
by Meidiana Ten 23 Feb '07

23 Feb '07

Hi there, I have problem in starting 2 SER at one machine at a time Can you guys show me how to start it.. I make 1 SER for home proxy (ser.cfg) and the other for SEMS (ser_sems.cfg) on same folder then I start ser.cfg with command #ser, it works well.. but when I try to start ser_sems.cfg with command #ser_sems, it doesn't work. It said it's unknown command Please show me where do I get wrong Thanx Regards, Meidiana ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

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Re: [Serusers] lookup_contacts() error after sd_lookup() from the speeddial module- latest cvs ottendorf
by TZieleniewski 23 Feb '07

23 Feb '07

2 5

[Users] ACK transaction
by Cseke Tamas 23 Feb '07

23 Feb '07

Hello, I'd like to relay sip calls from openserto another sip proxy (asterisk), and i have a problem with radius accounting, i 'd like to report acks, but openser doesn't send accounting packet on it. As i understand ack requests must match a transaction flaged to accounting. Config look like this, so i think all transaction is flaged. modparam("acc", "report_ack", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) route{ ... setflag(1); setflag(2); ... } Asterisk retransmit the 200 ok, so i think there is something mistake in the ack. as i read in rfc 3261, ACK must match the dialog, callid, from, to, fromtag, to, tag, and sequence number is relevant. and ack on a 200 ok, is always a new transaction, however i see branch parameter is 0 in the ack, it must start with the magic cookie z9hG4..., isn't it? I dont understand what could be the problem, why 200 ok is retransmitted, and why not openser send acctpacket to radius. Please someone take a look at the sip messages openser->asterisk INVITE sip:0619997400@127.0.0.1:5061 SIP/2.0 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Via: SIP/2.0/UDP 217.116.32.20;branch=z9hG4bKde6a.a93de1c7.0 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.22;branch=z9hG4bKvDD2OX3SfnAE6cR5;rport=1720 Max-Forwards: 69 User-Agent: PA168S V1.53.006 CFG0 From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest> Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 Contact: <sip:222@217.116.36.2:1720> CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 275 [sdp] asterisk->openser SIP/2.0 200 OK Via: SIP/2.0/UDP 217.116.32.20;branch=z9hG4bKde6a.a93de1c7.0;received=217.116.32.20 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.2;branch=z9hG4bKvDD2OX3SfnAE6cR5;rport=1720 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest>;tag=as37b57c9f Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 CSeq: 2 INVITE User-Agent: eWorld Com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0619997400@217.116.32.20:5061> Content-Type: application/sdp Content-Length: 206 [sdp] openser->asterisk <-- SIP read from 217.116.32.20:5060: ACK sip:0619997400@127.0.0.1:5061 SIP/2.0 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Via: SIP/2.0/UDP 217.116.32.20;branch=0 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.2;branch=z9hG4bKKysauOpwGdpnsmlS;rport=1720 Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Max-Forwards: 69 User-Agent: PA168S V1.53.006 CFG0 From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest>;tag=as37b57c9f Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 Contact: <sip:222@217.116.36.2:1720> Authorization: Digest username="222", realm="siptest", nonce="45dea111f92e1d1f9bf408a3996413168c304ee9", uri="sip:0619997400@217.116.32.20:5061", response="50419464515ca2771ec4b2bde7147219", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 Thanks any help, Tamas

2 1

[Users] Authenticate INVITEs
by Octavarium 23 Feb '07

23 Feb '07

I have enabled mysql authentication of REGISTER messagess, but i need that only INVITE mesagges from devices properly registered in OpenSER can go through and make calls, and that INVITE mesagges from devices NOT registered are dropped. How can i do this? do you have a sample script? Thanx a lot!

2 1

[Users] Error Handling with Parallel Forking
by Chris Heiser 23 Feb '07

23 Feb '07

Hello gurus, I'm attempting to do some rather crazy parallel forking and have found a few issues I'm hoping to solve. I thought I'd solicit some advice from the experts. Some background: Each device registers with its own unique information. No two devices share the same credentials. Since a random user might want their number sent to multiple phones, and maybe even a cell phone, I've done the following: if(method=="INVITE" && !isflagset(5)) { if(avp_db_load("$ru/username","fork")) { t_on_branch("1"); avp_pushto("$ru/username","$avp(s:fork)/g"); }; }; Then the branch route looks like: branch_route[1] { setflag(5); route(3); } The idea here is the fork avp contains one or more possible destinations, but we don't store a complete URI here. The branch route sends us back through the normal routing logic to lookup local users on the system, and maybe route something to an external PSTN gateway, or another proxy. The problem: I haven't found a way to gracefully handle the event when one of the possible destinations is a device that hasn't registered. Say we allow the user to have a soft client, so we add the username to the list of fork avps. If the device isn't registered, I have 2 options: 1) Use a default route 2) Detect that I should know who this is, and error out Either way, an error is produced, but I don't see a good way of detecting if we should push the 404, or generate a 404 back to the client. The polite method would be: Are there other branches still going? If yes, just exit this branch If no, relay, or generate an error back to the client So my question is how to do this last part. Am I crazy for even thinking about doing this in OpenSER? BTW, I'm using v1.1.1 --Chris

2 1

[Users] uri module question
by Klaus Darilion 23 Feb '07

23 Feb '07

Hi! I have problems understanding the uri module. As far as I understand I can map aliases to subscribers - e.g. my main identity is klaus@mydomain, but I also want to use nickname@cooldomain in From: header with a single SIP account (single subscriber) Thus I can use uri_db module and uri table to check if klaus@mydomain is allowed to use nickname@cooldomain. Thus, what do I enter in the uri table? The definition is: id | username | domain | uri_user | last_modified ----+----------+--------+----------+--------------- In which column do I have to put which values? thanks klaus -- Klaus Darilion nic.at

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