sr-users February 2008

sr-users@lists.kamailio.org

146 participants
260 discussions

Re: [OpenSER-Users] Mediaproxy Vs RTP Proxy

by Andy Smith

Hi Ali, there is a good description on the mediaproxy module doc page, near the bottom (nathelper). Should clear things up for you! :P http://openser.org/docs/modules/1.3.x/mediaproxy.html cheers Andy.

16 years, 9 months

[Serusers] Help needed with SER imprementations

by voip termination

Hello, I´m new to SER Express and I needed to use SER, on the following imprementations, but do not know which to use. I want to use this imprementations for free in network internet calling, calling card via asterisk (a2billing), and PSTN calls for all the network users. 1. I need to setup SER, to routes all my incoming and outgoing sip calls, (in network users to users) free sip calls 2. Routes all PSTN calls to Asterisk for billing via a2billing and then Asterisk will do the PSTN call terminations of the calls. 3. All the sip user to be authenticated by Asterisk / a2billing for the billing via their user name and password. 4. SER to use asterisk for the users voicemail and other pbx features. Can anyone please help me out? Looking to hearing from you. Regards, Henry -- ================================ "Never Let Go Of A Dream Untill You Can Wakeup And Make It Happens" ================================

16 years, 9 months

[OpenSER-Users] match src_ip

by Andreas Heise

Hi all, I search for a simple way to match src_ip from 192.168.1.6 - 192.168.1.11. Any ideas, the following aren't working :-( ... if(src_ip==192.168.1.[6-11]) if(src_ip==192.168.1.[6|7|8|9|10|11]) thanks, Andreas

16 years, 9 months

[OpenSER-Users] Openser, inbound SIP "trunks" and REFER

by Robby Dermody

All, I have a situation where I have a OpenSER server frontending a set of Asterisk boxes (call them box A1 and A2). Calls come in and are routed to one of the boxes (after doing some custom HTTP lookups using a custom perl module, rewriting the RURI, etc). This works fine, however at some point, the call will most likely need to be transferred from one internal Asterisk box to the other Asterisk box (e.g. A1 to A2). At this point, what we have set to happen is that A1 would send a REFER up through the proxy, the proxy caches an auth token record from info in the outgoing REFER and writes the token ID into the Refer-To header (along with its own address as the recipient) as the REFER goes out to the SIP trunk, and then when the subsequent invite is made from the sip trunk (to something like route_34983948-34343@<sipProxyIP> with that 34983948-34343 being the token ID) the SIP proxy will look up that auth token, rewrite the RURI with the cached info and direct the call to the appropriate internal asterisk box...A2 in this case. So that all is done and would normally work. However our SIP provider says that it only provides "inbound DID service" and thus do not have their dialplan set up to handle REFERs with custom URIs (even if it is back down to the same openser proxy, as it always is in our case). So looking at this, I have two questions: 1. Could resolving this problem just be as simple as finding a new SIP DID provider? I am a bit worried that providers may consider use of REFERs in any way as "SIP termination" (e.g. outgoing service) and either not office REFER support or charge accordingly for outbound access, even though this is clearly still inbound. 2. Those Asterisk boxes in our environment are behind a NAT, so our openser proxy does employ nathelper/rtpproxy. Given this, another alternative I see is for the openser proxy to intercept the REFER for the DID->A2 message, craft an INVITE for proxy->A2, negotiate the conversation, and reconstruct the internal RTP leg between it and A2 instead of A1. This way, the SIP trunk is none the wiser and does not even know this transfer has happened, it just gets some dead air for a second or two and everything is done behind the scenes. I know that a SIP proxy shouldn't inject new SIP messages into the stream, and the REFER way seems to be the most RFC-compliant and technically sound method. However, there is the real-world threat in my mind that it may not be very well supported among inbound DID providers, so I am interested in the community's opinion on the options. Thanks, Robby

16 years, 9 months

[OpenSER-Users] Route calls to PSTN Gateway

by Ali Jawad

Hi All I have successfully setup OpenSer with a mysql backend. I am able to make calls using sip phones between registered users. What I need to do know is routing calls that start with 00 to my PSTN gateway. But first I need to setup OpenSer to route calls to my PSTN ..! I am using OpenSer1.3 and I have checked sipwise.com but honestly I prefer changing my own config file. But from sipwise.com I concluded that I need the config below But are the nathelper and mediaproxy compiled by default or do I have to recompile with special options ? Other than that where do I set the IP of my PSTN gateway ? Any suggestions or help would be welcome thanks. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket","/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics","/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics","/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so"

16 years, 9 months

[OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW

by Ali Jawad

Hi All I am using OpenSER as a proxy to make outbound calls my config is very simple if the number dialled is not an openser account route it to the PSTN gw. if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } } else { # probably a call to pstn.... route(2); return; } and route[2] { # pstn handling, simply route out to pstn. sethostport("xx.xx.xx.xx:5060"); route(1); } The problem is that once the SIP packet arrives at the PSTN GW it does NOT have the correct TO: set. Therefore the call does not get routed . In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP 192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068. Record-Route: <sip:193.237.226.252;lr=on>. From: "ssafass" <sip:ali@jabber.splendor.net>;tag=f36d6608. To: "009613041705" <sip:009613041705@jabber.splendor.net>;tag=GR52RWG346-34. Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" <sip:PSTN.GW.IP.IP:5060>. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0. I did a siptrace on the interface of the SIP proxy http://pastebin.com/d56426d63 <http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63> This is my config: http://pastebin.com/m128ca16e <http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e>

16 years, 9 months

[OpenSER-Users] Not able to connect Application Server to the SEAS module of OpenSER.

by Rajat Dudeja

Hi Folks, My openser is running on SCTP. I'm using SEAS module. I am using a simple client socket (TCP Socket) program as Application Server(AS) to connect to SEAS module. As is written to connect to port 5080 of server. But it does not connect as is evident from the line (openser' console log): Feb 1 17:30:47 [12031475] INFO:seas:dispatcher_main_loop: polling [2 ServSock] [1 pipe] [0 App Servers] [0 Uncomplete AS] I'm sending a REGISTER message from a SCTP Client to OpenSER on port 5070. I'm expecting the same REGISTER to be passed to the AS. I'm using the following openser.cfg file : ####### Global Parameters ######### #debug=3 #log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=1 dns=no /* uncomment the following lines to enable debugging */ debug=9 #fork=no log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "//etc/openser/tls/user/user-cert.pem" #tls_private_key = "//etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "//etc/openser/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=SCTP:157.227.110.27:5070 ####### Modules Section ######## #set module path #mpath="//lib/openser/modules/" mpath="/home/ops/openser/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" #Rajat-Jiten added 01.02.08 loadmodule "seas.so" # ----------------- setting module-specific parameters --------------- #Rajat-Jiten added 01.02.08 # ----- seas params ----- modparam("seas", "listen_sockets","157.227.110.27:5080") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/abc/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- modparam("usrloc", "db_mode", 0) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## route{ if(!as_relay_t("app_server")){ t_reply("500","Application Server error"); } } =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

16 years, 9 months

[OpenSER-Users] openser+opendiameter integration problem----->KINDLY PAY ATTENTION

by bhargav.b＠renovau.com

hello everybody, i have some problem regarding the integratoion b/w open diameter and openser for clients authorization. i have recently installed openser and opendiameter as needed.both are working fine independently.but for the integration i need somebodies kind help.i have run all the sample examples of both. even my softphone[xlite]is doing fine with openser.can anybody please tell me how can i configure the openser.cfg to support diameter? please also sujjest me regarding avp's creation for the diameter+openser support. thank you very much bhargav.b bhargav.b(a)renovau.com

16 years, 9 months

[OpenSER-Users] openser+opendiameter integration problem----->KINDLY PAY ATTENTION

by bhargav.b＠renovau.com

hello everybody, i have some problem regarding the integratoion b/w open diameter and openser. i have recently installed openser and opendiameter as needed.both are working fine independently.but for the integration i need somebodies kind help.i have run all the sample examples of both. even my softphone[xlite]is doing fine with openser.can anybody please tell me how can i configure the openser.cfg to support diameter? please also sujjest me regarding avp's creation for the diameter+openser support. thank you very much bhargav.b bhargav.b(a)renovau.com

16 years, 9 months

Re: [OpenSER-Users] Reg. Proxy domains

by Padmaja

Hi.. That is the whole issue,,, The proxy 1 and proxy2 do not know each other. How can I configure openser so that it recognizes proxy1 as another proxy and sends/receives invites? I do not want any of the proxies acting as out bound proxy. Both should be primary proxies for their respective domains and yet they should be able to recognize the other proxy and forward the requests to it. In other words, what I am asking how do the proxies talk to each other? Thanks, Padmaja ----- Original Message ----- From: raviprakash sunkara To: Padmaja Sent: Thursday, January 31, 2008 1:55 PM Subject: Re: [OpenSER-Users] Reg. Proxy domains Hello Padmaja Akka, IS urs Proxy1 sending the INVITE to Proxy2.. Can u trace by using ngrep or trace and Send to u .... On Jan 31, 2008 12:39 PM, Padmaja <padmaja.rv(a)vodcalabs.com> wrote: Hi all, Please see the situation below: UA1 ----------------------> Proxy 1(not openser)---------------------- Proxy 2(openser)<----------------------UA2 UA1 is registered to proxy 1 and UA 2 to openser. I wish to make a call from UA 1 to UA 2. Proxy 1 and proxy 2 are in the LAN but do not know of each other. Is there a way we can configure the openser to send/receive the invites to proxy 1? I think the calls will be handled on domain basis, so openser will treat all the UAs registered to it as local and allow them to call each other. But how to make it allow inter domain calls? Please let me know. Thanks, Padmaja _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

16 years, 9 months

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sr-users February 2008