sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

RE: [Serusers] Looking for Call termination service provider

by John Kim

We are in USA and we are targeting the calls terminated in South Korea. -----Original Message----- From: ram [mailto:talk2ram@gmail.com] Sent: Thursday, August 10, 2006 11:27 AM To: John Kim Subject: Re: [Serusers] Looking for Call termination service provider Hi where are you from ? I can asissts you Ram On 8/10/06, John Kim <johnk(a)koreanet.us> wrote: Hi, I am looking for a VoIP wholesaler who will allow us to operate our own SIP server infrastructure while we can forward all PSTN destined calls to their gateway. I really didn't think that it would be hard to find one but I have not been successful. I have checked so far with iCallGlobe.com and Deltathree.com <http://Deltathree.com> . Can any of you recommend us some VoIP wholesaler with quality service and competitive rate who allows this business setup? Below is logical business relation setup: Our subscriber <--->ATA setup to connect sip.ours.com<----->PSTN gateway of wholesaler.com<--->PSTN Basically we are looking for good " wholesaler.com <http://wholesaler.com> ". Any comments will be appreciated. John K. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

18 years, 3 months

[Serusers] ser SDP rewrite problem, rtpproxy is not working !!!

by Sebastian Gabris

Hi, got also this problem, did you find a solution? Sebastian hello there, I am working on SER with two network interfaces. EP1--------------->(172.16.16.160) SER (192.168.5.1) -------------> EP2 what happens, SER receives invite from EP1, it reinitiates invite for EP2. now what it does is that it writes the external IP i.e. 172.16.16.160 in the SDP contact info, where as it should be the internal one i.e. 192.168.5.1 ... now I want to overwrite the SDP contact info in that invite. I have got two options to do it one option is to use force_rtp_proxy("ei", "192.168.5.1") or use sdp_mangle_ip() function of mangler.so . but they are not working for me. when I play around with force_rtp_proxy function rtpproxy get crashed and sdp_mangle_ip() is not behaving properly ...... whats your experience ? I have used ser-0.8.14 and now using ser-0.9.0 with cvs rtpproxy. I am running rtpproxy like rtpproxy -s udp:192.168.5.1 -l 172.16.16.160 and I am using this line in ser.cfg modparam("nathelper", "rtpproxy_sock", "udp:192.168.5.1") I am really stucked please help !!!! thanking you regards, -- Atif ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

18 years, 3 months

[Users] extract substring?

by zm23

Hi all, Is there a way to extract a substring from uri and save that value? I see subst function in textops module but that is only to substitute. thanks. -- Zahid Mehmood CUIT Network Systems

18 years, 3 months

[Serusers] Problem with incoming calls

by Sebastian Gabris

Hi everybody, Like told in the subject i got problems with incomnig calls, SIP signalling works but RTP doesnt. When I make a call from my UA to the VoIP-Provider all works well, sip & rtp. I start rtpproxy in bridged mode: rtpproxy -l publicIP/privateIP. I use this scenario: UA(private IP) <-> (privateIP) SER (publicIP) <-> VoIP-Provider i try to figure out where my mistake is in this config file: route[1] { if(dst_ip == 10.245.104.68) { force_rtp_proxy("","193.77.54.68"); } else { force_rtp_proxy("","10.245.104.68"); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { if(dst_ip == 10.245.104.68) { force_rtp_proxy("","193.77.54.68"); } else { force_rtp_proxy("","10.245.104.68"); }; }; } By the way, is there a option to view rtpproxy debug info? Thanks in advance! Sebastian ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

18 years, 3 months

[Users] I can`t make a calling by using xlite at the client. Please help me...Please

by Ferianto siregar

Dear all, I would like to say thanks for everybody help and kind-hearted who always give me some helps to solve my problem. Thank you very much. Now, I have a problem in my openser. I can`t make a calling from client using the xlite. It is meant that the client can not make a call by using xlite ( It can not connect to server) . In xlite screen, it always shows "403 forbidden" message. Does anybody can give me a suggesstion what is meant? Where the problem comes from? Please help me ...Please. For the information: 1. The client use IP 202.95.149.2 2. This is the log message that I have got from xlite diagnostic: SEND TIME: 3836510 SEND >> 202.95.149.250:5060 REGISTER sip:pcr.ac.id SIP/2.0 Via: SIP/2.0/UDP 202.95.149.40:5060;rport;branch=z9hG4bKC098998FE862469BBE0A3F218390D2B1 From: aldi <sip:123@pcr.ac.id>;tag=4145118495 To: aldi <sip:123@pcr.ac.id> Contact: "aldi" <sip:123@202.95.149.40:5060> Call-ID: AF8EEFDE9AD94CC5B910233C4AFA09BF(a)pcr.ac.id CSeq: 5795 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105x Content-Length: 0 RECEIVE TIME: 3836512 RECEIVE << 202.95.149.250:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 202.95.149.40:5060;rport=5060;branch=z9hG4bKC098998FE862469BBE0A3F218390D2B1 From: aldi <sip:123@pcr.ac.id>;tag=4145118495 To: aldi <sip:123@pcr.ac.id>;tag=98a2bfb09e790c090877fa710657efb6.0d98 Call-ID: AF8EEFDE9AD94CC5B910233C4AFA09BF(a)pcr.ac.id CSeq: 5795 REGISTER Server: OpenSer (1.1.0-tls (i386/linux)) Content-Length: 0 Warning: 392 202.95.149.250:5060 "Noisy feedback tells: pid=4439 req_src_ip=202.95.149.40 req_src_port=5060 in_uri=sip:pcr.ac.id out_uri=sip:pcr.ac.id via_cnt==1" 3.What is the meaning : " 2(4454) is_local(): Realm 'pcr.ac.id' is not local"? I got it when I run "openser" command. This is the message that I got: get_hdr_field: cseq <CSeq>: <1497> <REGISTER> 2(4454) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(4454) parse_headers: flags=200 2(4454) DEBUG: get_hdr_body : content_length=0 2(4454) found end of header 2(4454) find_first_route: No Route headers found 2(4454) loose_route: There is no Route HF 2(4454) is_local(): Realm 'pcr.ac.id' is not local 2(4454) DEBUG: add_param: tag=888227406 2(4454) DEBUG:parse_to:end of header reached, state=29 2(4454) DBUG:parse_to: display={123}, ruri={sip:123@pcr.ac.id} 2(4454) is_local(): Realm 'pcr.ac.id' is not local 2(4454) parse_headers: flags=ffffffffffffffff 2(4454) check_via_address(202.95.149.2, 202.95.149.2, 0) 2(4454) DEBUG:destroy_avp_list: destroying list (nil) 2(4454) receive_msg: cleaning up 4(4460) udp_rcv_loop: probing packet received from 202.95.149.2 50195 5(4461) udp_rcv_loop: probing packet received from 202.95.149.2 50195 2(4454) udp_rcv_loop: probing packet received from 202.95.149.2 50195 4.This is the contain of mysql: mysql> select * from aliases; +----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+| username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | user_agent | socket | methods |+----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+| 123 | pcr.ac.id | sip:coba@pcr.ac.id | NULL | NULL | 2006-08-09 02:53:59 | 1.00 | The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything | 42 | 1970-01-01 07:00:00 | 128 | OpenSER Server FIFO | NULL | NULL || 456 | pcr.ac.id | sip:feri@pcr.ac.id | NULL | NULL | 2006-08-09 06:57:28 | 1.00 | The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything | 42 | 1970-01-01 07:00:00 | 128 | OpenSER Server FIFO | NULL | NULL |+----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+2 rows in set (0.00 sec) I do hope anybody can help me. I have tried to solve this problem 2 days but I can`t. Please help me..please.. Regards with cheers, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1ï¿½/min. --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

18 years, 3 months

Re: [Serusers] presence_b2b UA

by Vaclav Kubart

Hi Curt, are presence subscriptions enough? PA module is prepared to generate internal (QSA) subscription when it is processing SUBSCRIBE request. But it still can not handle responses - this will be done soon. I think that this will be what do you need. I hope that I will have time for it and finish it till the end of August. Vaclav On Wed, Aug 09, 2006 at 02:18:07PM -0500, Curt Moore wrote: > Hi Vaclav. > > Thank you for your reply. > > This sounds very close to what I'm trying to accomplish for presence. > I have 1 SER server setup as the main proxy/registrar. If a SUBSCRIBE > message comes into this server, it is forwarded over to a second SER > server which maintains the watcher info, a presence server. This > currently works perfectly. > > What I'd like to happen is for the B2B UA on the presence server to > generate a new SUBSCRIBE message and send it out to an Asterisk > server, essentially having the SER presence server subscribe to the > presence of the SIP user on the Asterisk server. If the presence > changes on the Asterisk server, it will send a NOTIFY back to the > presence SER server. This should then, as you mention, trigger a > notification through QSA back to any watchers watching this particuar > resource. > > My confusion is how to get the outbound SUBSCRIBE generated through > QSA and sent to the Asterisk server from the presence server after the > initial inbound SUBSCRIBE comes into the SER presence server. > > How would I get the SER server to generate the outbound SUBSCRIBE? > > Thanks! > -Curt > > On 8/9/06, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >Hi, > >presence_b2b processes INTERNAL subscriptions (through QSA - internal > >API for status querying) and generates SIP subscriptions according to > >them (now only presence subscriptions). Each arriving NOTIFY triggers > >notification through QSA. > > > >It works quite good with one SER as resource list server and another as > >presence server. > > > > Vaclav > > > >On Wed, Aug 09, 2006 at 12:01:59PM -0500, Curt Moore wrote: > >> Hello all. > >> > >> I have been experimenting with the ser-0.10.99-xxx branch(es) and have > >> a question about the presence_b2b module. > >> > >> I realize that the module is not yet fully complete but according to > >> the documentation, when finished, the module should be able to take an > >> inbound SUBSCRIBE request and then create a new SUBSCRIBE request > >> outbound to another SIP entity or multiple SIP entities, ie Asterisk > >> servers, hence the back to back functionality. What is the status of > >> this functionality? Is anyone currently working to complete this > >> functionality? I would be interested in helping complete or > >> sponsoring the completion of this functionality. > >> > >> My intent is to have a SER server act as the central watcher database > >> server and have all SUBSCRIBE and NOTIFY requests channeled through > >> this SER server. This SER server will then generate new SUBSCRIBE > >> requests out to each server in a farm of Asterisk servers and the > >> Asterisk servers will then send NOTIFIES back to the SER watcher > >> server. The watcher server will then take care of sending the NOTIFY > >> messages to the correct watcher(s), according to the data in its > >> watcher database. > >> > >> Is anyone currently doing anything similar to this using a different > >> method or is using the yet to be completed B2B UA the correct/only way > >> to accomplish this? > >> > >> Thanks, > >> -Curt > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >

18 years, 3 months

[Serusers] Accounting Calls not by extension

by Nhadie Ramos

Hi All, I'm doing accounting via the usage of per extension. Can i also do it when a client is a gateway and would forward calls to my SER? e.g. from their gateway, the will append everything infront of their dialled digits with 12345#, then forward it to my SER. then i'll strip the 12345# then forward it to my gateway provider. how can i account anything that has 12345# or can I account it via the IP address of their gateway? TIA Regards, nhadie

18 years, 3 months

[Serusers] to use SER stateless proxy..................

by dillip parida

hi all, I am newbie to SER. actually I want to use SER as stateless proxy.. i will explain with a example......user agent A and User agent B are two SIP phones. if A calls , B's number initially it reaches at SER, then SER route the call stalelessly to particular URI. BUT I DONT NEED THAT USER 'A' and USER 'B' SHOULD NOT GET REGISTERED AT SER. without registering at SER we need to route the call to that particular URI....i mean to that user B..... please help me.......i have attached the ser.cfg for ur refrence.

18 years, 3 months

Re: Re: [Users] two phones ringing

by Mark Kent

>> By the way, when you answer very quickly, did the other phone ring at >> least once before? The phone I answered rang once, the slow phone did not. If they happen to both ring at the same time then it works, it's just the times that the slow phone lags that causes the problem. Thanks, -mark

18 years, 3 months

Re: [Serusers] Multihomed SER & NAT

by Sebastian Gabris

Thanks for the response, will try it and see what happens. :-) While waiting for help, i tried a new config, without fix_nat_contact. It works (Audio & signalling) for outgoing calls.(private -> public) Contact header has a private IP but SDP information is correct due to RTPProxy. At incoming calls the force_rtp_proxy command didn't change anything, meaning that SDP Information is not correct -> no audio, only signalling. I tried to start rtpproxy without options and in bridged mode. Also in bridged mode i tried both possibilities -l private/public & -l public/private, but it didn't influence the outcome. Perhaps someone could help me out on this. :-) Here are the changes to my config. route[1] { if(dst_ip == 10.x.x.x) { force_rtp_proxy("","193.x.x.x"); }; if (dst_ip == 193.x.x.x) { force_rtp_proxy("","10.x.x.x"); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { if(dst_ip == 10.x.x.x) { force_rtp_proxy("e"); log(2,"RTP_PROXY: Reply-Route1 dst 10.x.x.x\n"); }; if (dst_ip == 193.x.x.x) { force_rtp_proxy("i"); log(2,"RTP_PROXY: Reply-Route1 dst 193.x.x.x\n"); }; }; } fix_nated_contact() is meant for changing the IP address of the Contact > header when SER has a public address and the UA has a private (i.e. > behind NAT). This means that it will rewrite Contact to contain the > source ip address:port found in the UDP/TCP header. > This is probably correct, because I assume the UA cannot be reached > from the outside anyway. If you record-route, subsequent requests > should go through your ser. > > If you have problems with responses not being returned, have a look at > Via headers as they control how answers to i.e an INVITE are routed > back to the UA. You may want to have a look at the rr module double > record route parameter to make sure that SER creates one record-route > for the internal and one for the external interface. > g-) > > Sebastian Gabris wrote: > After reading a bit more mails, I came to two possible problems: > 1. fix_nated_contact() needs a registrar to work properly > I dont use a registrar, since i register at the VoIP - Provider. > or > 2. fix_nated_contact() is not ment to be used on this scenario > Any ideas on this? > Sebastian > > > Hi everybody, > need some help on nattraversal. > i use a multihomed SER, one private IP & one public IP. > MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) > If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address. > Instead o writing down his public IP, SER writes his private IP. > Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information. > After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :) > Here are the important bits of my config: > fork=yes > log_stderror=no mhomed=1 > listen=10.x.x.x > listen=193.x.x.x > loadmodule "/usr/lib/ser/modules/nathelper.so" > modparam("usrloc", "db_mode", 0) > 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 > if (nat_uac_test("3")) { > if (method == "REGISTER" || ! search("^Record-Route:")) { > log(2,"LOG: Someone trying to register from private IP, rewriting\n"); 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 > }; > }; > ... and so on. > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > _____________________________________________________________________ > Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! > http://smartsurfer.web.de/?mc=100071&distributionid=000000000071 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

18 years, 3 months

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sr-users August 2006