sr-users October 2003

sr-users@lists.kamailio.org

105 participants
230 discussions

by Stephen Miles

Thanks for that, I'll get the source from the CVS and give it a go. -----Original Message----- From: Andrei Pelinescu-Onciul [mailto:pelinescu-onciul@fokus.fraunhofer.de] Sent: Tuesday, 28 October 2003 10:05 a.m. To: Stephen Miles Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] RTP Proxy and SER On Oct 28, 2003 at 09:54, Stephen Miles <smiles(a)callplus.co.nz> wrote: > Hi Andrei, > > I have the nathelper module being loaded but am still getting an error. > Here is my log, can you shed any light on this for me? Yes, force_rtp_proxy exists only in the nathelper module from ser unstable (cvs HEAD). Sorry I didn't notice the part about using ser debian packages (which implies you use ser stable). You could try getting ser unstable sources and compiling it, however right now we are doing a lot of changes and the current cvs might be broken. It is probably safer to use a version from 2 weeks ago. Andrei

20 years, 6 months

[Serusers] How does SER's Stateful Proxy works?

by Dharani Narayanan

Hello I went through few modules of SER regarding the session maintenance. When i walked through the Transaction Module, I came to know that SER provides Stateful Proxy. I have few questions regarding this: 1. How does SER maintains session? i.e., Does it maintain any state table similar to file/socket descriptor? 2. In the case of "dialog" interaction between clients are identified by callID and routed through the Record-Route. Does Record-Route traces the session? If so where it is maintained? Will this become a overhead in header while routing all the details? Does the client feel efficient in performance? 3. How does SER knows whether the session is disconnected or logged off? And under what situation does the SER stores the offline messages? 4. Can you provide me the total architecture of how the stateful proxy is designed and works? Can you please update me for the above mentioned questions. Thanks Dharani. __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

20 years, 6 months

[Serusers] ser does not respond to messages with NATed address

by Yang Xiang

Hello, I have a NATed network behind a cisco PIX firewall as follows: client --- SER 1-----PIX (NAT) --------SER 2 The client has a private address 192.168...., and is NATed to a public IP. Because PIX has a bulit-in SIP-application-gateway, the IP of the client in message body will be transformed to his public IP by firewall.. However, what most strange is SER2 doesn't respond to messages from SER1. -------------- #################### U 153.96.51.2:5060 -> 153.96.52.2:5060 INVITE sip:6091@153.96.52.2:5060 SIP/2.0..Record-Route: <sip:6091574@153.96.51.2;ftag=781a5806-7443-4c67-b8a6-4ae6f9b64bc8;lr=on>..Max-Forwards: 9 . .Record-Route: <sip:6091@192.4.3.2;ftag=781a5806-7443-4c67-b8a6-4ae6f9b64bc8;lr=on>..Via: SIP/2.0/UDP 153.96.51.2;branch=z9hG4bK75cf.a543.0..Via: SI P/2.0/UDP 192.4.3.2;branch=0..Via: SIP/2.0/UDP 153.96.51.77:9779..From: "xia" <sip:xia@nd.fhg.de>;tag=781a5806-7443-4c67-b8a6-4ae6f9b64bc8..To: <s ip:6091@nd.fhg.de>..Call-ID: 9947a08e-d8c5-47ef-90a3-cdd10f3b143c@153.96.51.77..CSeq: 1 INVITE..Contact: <sip:153.96.51.77:9779>..User-Agent: W indows RTC/1.0..Content-Type: application/sdp..Content-Length: 457....v=0..o=yang-mobil 0 0 IN IP4 153.96.51.77..s=session..c=IN IP4 153.96.51.77..b=CT :1000..t=0 0..m=audio 61052 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7 221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtp map:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16.. ---------------------------------------------- (originally the client has an IP 192.168.44.33 and it has been transformed to 153.96.51.77 by firewall as seen above) SER2 gets all messages repeatedly from SER1 but does nothing as if it hadn't seen any messages. (no reply sent, no log written) If I change the client to another network with not NATed adress, it works fine! SER1's config: --------------------------- ... if (uri=~"^sip:[0-9]*@.*"){ # PSTN call after the authorization. Authorization has been done in bolck route[2] and [3] if (uri=~"^sip:00[0-9]+@.*"){ sl_send_reply("699", "Sorry, international call not allowed"); break; }; log(2, "Forward to ser2 \n"); rewritehostport("153.96.52.2:5060"); forward("153.96.51.2"); break; }; ... ----------------------- And SER2's config: # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ sl_send_reply("100","msg received, trying"); ### (not any reply was sent as msg arrived!) if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","too many hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "msg too big"); break; }; record_route(); loose_route(); if (search("(c|C)ontact:.*192\.168.*")) { fix_nated_contact(); if (method=="INVITE") { fix_nated_sdp("2"); }; }; if (!src_ip==10.20.0.2) { log(3, "Forward to inn. ser\n"); forward("10.20.0.2"); break; }; # if the request is for other domain use UsrLoc if (!t_relay()) { sl_reply_error(); }; } --------------------------------------------------- Any idea? Thanks Yang

20 years, 6 months

[Serusers] voicemail installation

by Chen Youcheng

Hallo, ich habe iptel ser instaliert. Es funktioniert gut. ich moechte jetzt Voicemail integrieren. unter http://www.iptel.org/ser/doc/prerelease/x1294.html ist die Beschreibung: First, you need to compile Ser with voicemail support. Thatfor, you must edit Ser's Makefile.defs file and uncomment the line with '-DVOICE_MAIL' and '-D_TOTAG'. Es gibt es keine '-DVOICE_MAIL' and '-D_TOTAG'. Kanst du mir mal helfen? dayong1973 __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

20 years, 6 months

[Serusers] Does SER support load balancing on the server-side?

by Madhuri garigipati

Hello, I have a question on load balancing. Does SER support load balancing on the server-side? If yes, how does it provide load balancing? What is the mechanism used? Thanks, Madhuri. __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

20 years, 6 months

Re: [Serusers] multiple users registering problem

by Jiri Kuthan

There may be lot of reasons why the old contact disappears. A typical reason is that the old contact expires after its expiration period. If you don't think that is the case, send us your message dumps. regards, -jiri At 03:01 AM 10/22/2003, Annie Sasidar wrote: >Hi, > I am using Windows messenger 4.7 for testing ser. I have a linux box as a >stateless proxy. Only a single user gets registered. When i register a new >user, the old record seems to be deleted. When i do a serctl ul show i get >only the last user registered. What should i do? >Any tips will be greatly appreciated. > >Thanks in advance, >Annie > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

20 years, 6 months

Re: [Serusers]SIP Gateway

by Director General: NEFACOMP

Hi, I have received it but I couldn't reply because I am not in town now. I will check tomorrow to see if I can help. Thanks Emery ----- Original Message ----- From: "Sesha B" <sesha(a)iic.com> To: "Director General: NEFACOMP" <dg(a)nefacomp.net> Sent: Monday, October 27, 2003 17:05 Subject: RE: [Serusers]SIP Gateway > I'm sorry to trouble you again and again, but did you get the mail I sent > you earlier about my setup? Please let me know if you found anything wrong > with the setup or the config? Thank you very much. > > Sesha > > -----Original Message----- > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > Sent: Friday, October 24, 2003 7:06 AM > To: sesha(a)iic.com; serusers > Subject: Re: [Serusers]SIP Gateway > > > You asked: > >> Does the mediatrix be assigned a number too? > Can you explain? > Normally each port on the Mediatrix is like a UA (like Xten, ... [I might be > wrong!!]) > That is why, the account you tell it to use should be created in SER first. > > NGREP can be found at http://ngrep.sourceforge.net/ > After installing it you may use [>>>>>ngrep port 5060] to watch the traffic > on port 5060. > > When you get the dial tone, what do you do with it? Normally you should be > able to dial a number to call. > For example if the Xten software is running on your PC and registered to use > 500 as main alias, after getting the dial tone (with the phone connected to > the Mediatrix) you may dial 500 without hanging up and your UA will start > ringing on the PC. > > Can you tell me in more details how your devices are interconnected? > > > Thanks > Emery > ----- Original Message ----- > From: "Sesha B" <sesha(a)iic.com> > To: "Director General: NEFACOMP" <dg(a)nefacomp.net> > Sent: Thursday, October 23, 2003 16:58 > Subject: RE: [Serusers]SIP Gateway > > > > The analog gateway ID is in readonly mode in my MIB and it is set to 5 by > > default. I have disabled the redirection and I get the dialtone after two > > rings. And with the original configuration I have, I get the engage tone > > after two rings. I have one question! Does the mediatrix be assigned a > > number too? Also, can you please let me know how do use ngrep to watch the > > registration headers on the SER registrar? Thank you. > > > > Sesha > > > > -----Original Message----- > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > Sent: Thursday, October 23, 2003 10:01 AM > > To: sesha(a)iic.com > > Subject: Re: [Serusers]SIP Gateway > > > > > > I also have another question about your device: > > analogGatewayID (.1.3.6.1.4.1.4935.1.1.1.4.1.2.0) = 5 > > > > The above line says that your gateway ID is 5. It seems it is different > from > > mine. Mine is 2. > > Possibly I won't be of much help to you. But we can always try. > > > > Can you remove the redirection and see if it works? > > gatewayPort1RedirectEnable (.1.3.6.1.4.1.4935.1.1.1.4.1.14.10.2.0) = 0 > > ////// Just disable the redirection to see if that helps > > gatewayPort1RedirectToAddress (.1.3.6.1.4.1.4935.1.1.1.4.1.14.10.3.0) > = > > x.x.x.x > > gatewayPort1NetToPstnSourceFilter > > (.1.3.6.1.4.1.4935.1.1.1.4.1.14.10.4.0) = 24.105.144.220 > > > > For the remaining parts, I don't see any difference with my configurations > > (except that you are using country codes while me I have disabled them) > > > > > > Thanks > > Emery > > ----- Original Message ----- > > From: "Sesha B" <sesha(a)iic.com> > > To: "Director General: NEFACOMP" <dg(a)nefacomp.net> > > Sent: Thursday, October 23, 2003 14:32 > > Subject: RE: [Serusers]SIP Gateway > > > > > > > I'm attaching my getwalk from the branch voipgateway, which includes > > > everything. The phone I was mentioning earlier is the SIP phone into > which > > > I'm trying to dial from the regular phones on the PBX. I've the > mediatrix > > on > > > 4 ports on the PBX. Thank you. > > > > > > -----Original Message----- > > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > > Sent: Thursday, October 23, 2003 4:40 AM > > > To: sesha(a)iic.com; serusers > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > What phone? > > > Can you send me the GetWalk generated by your Unit? > > > > > > > > > Thanks > > > Emery > > > ----- Original Message ----- > > > From: "Sesha B" <sesha(a)iic.com> > > > To: "Director General: NEFACOMP" <dg(a)nefacomp.net>; <serusers(a)lists.iptel.org> > > > Sent: Thursday, October 23, 2003 00:34 > > > Subject: RE: [Serusers]SIP Gateway > > > > > > > > > > I have these parameters already set. I also restarted mediatrix after > > > > > setting these MIB variables but I can see only the SIP phone > registering > > > > with the SER in the ser diagnostics but I dont see the SIP gateway's > > > address > > > > at all !!! > > > > > > > > -----Original Message----- > > > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > > > Sent: Wednesday, October 22, 2003 10:47 AM > > > > To: sesha(a)iic.com; serusers(a)lists.iptel.org > > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > > > > I will send instructions for the first port and you will be able to do > > the > > > > same for the other ports. > > > > > > > > In the provisioning MIB, find the key: > > > > sipMib -> sipUAGroup > > > > Start by configuring the SER address: > > > > sipMib -> sipUAGroup -> sipUAServerGroup -> sipUAServerStaticGroup > > > > and configure the parameters therein. Such as Registrar Host, Port > > ... > > > > > > > > Second, configure the port 1 as follows: > > > > sipMib -> sipUAGroup -> sipUA1Group > > > > configure the main alias entry which should be the account created > > on > > > > the SER server for that port > > > > you may even configure other things if needed such as other > aliases, > > > > .... > > > > For other ports, just replace # in sipUA#Group by the correct port > > number > > > > (2, 3 or 4) > > > > > > > > Also, you may need to enable some options in the provisioningMib key. > > > > > > > > > > > > Thanks > > > > Emery > > > > ----- Original Message ----- > > > > From: "Director General: NEFACOMP" <dg(a)nefacomp.net> > > > > To: <sesha(a)iic.com>; <serusers(a)lists.iptel.org> > > > > Sent: Wednesday, October 22, 2003 15:53 > > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > > > > > I was asking the firmware version (the software installed on the > > 1204). > > > > Not > > > > > the Unit Manager. > > > > > > > > > > I am going to send you instructions on how to set up accounts in the > > > unit. > > > > > These are accounts the unit uses in registering with the SER server. > > > > > Note that you will need to configure the same accounts within SER. > > > > > > > > > > > > > > > Thanks > > > > > Emery > > > > > ----- Original Message ----- > > > > > From: "Sesha B" <sesha(a)iic.com> > > > > > To: "Director General: NEFACOMP" <dg(a)nefacomp.net>; > > <serusers(a)lists.iptel.org> > > > > > Sent: Wednesday, October 22, 2003 14:55 > > > > > Subject: RE: [Serusers]SIP Gateway > > > > > > > > > > > > > > > > Hi, > > > > > > I'm using Unit Manager Express version 1.1.1.5 to configure > > Mediatrix > > > > > 1204. > > > > > > I have configured it to use SIP. I am confused about setting up > > > accounts > > > > > on > > > > > > Mediatrix. Is there a way to set up accounts on mediatrix? I'm > > > testing > > > > it > > > > > > with one SIP phone. > > > > > > > > > > > > -----Original Message----- > > > > > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > > > > > Sent: Wednesday, October 22, 2003 4:55 AM > > > > > > To: sesha(a)iic.com; serusers(a)lists.iptel.org > > > > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > > > > > > > > > > I used Mediatrix 1204. > > > > > > Have set up all the accounts that Mediatrix should use in > > registering > > > to > > > > > > SER? > > > > > > What software version are your Mediatrix using? You sometimes need > > to > > > > > > upgrade your firmaware. > > > > > > Also, the same Mediatrix can be H.323 or SIP. Are you sure that > > yours > > > is > > > > > > using SIP? > > > > > > > > > > > > > > > > > > Thanks > > > > > > Emery > > > > > > ----- Original Message ----- > > > > > > From: "Sesha B" <sesha(a)iic.com> > > > > > > To: <serusers(a)lists.iptel.org> > > > > > > Sent: Monday, October 20, 2003 23:02 > > > > > > Subject: RE: [Serusers]SIP Gateway > > > > > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > I'm using a SIP gateway (Mediatrix 1204) on 4 ports on PBX. I am > > not > > > > > able > > > > > > to > > > > > > > register the gateway on the SIP server. Is there a way I check > the > > > > logs > > > > > on > > > > > > > the SER to see if it is really registering or not? And can > someone > > > > > please > > > > > > > suggest why is the SIP Gateway not registering with the SER? > Thank > > > you > > > > > > very > > > > > > > much. > > > > > > > > > > > > > > _______________________________________________ > > > > > > > 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 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >

20 years, 6 months

[Serusers] Click to dial

by Teppo Lunnas

Hello, Do you happen to know if Messenger 5.0 supports click-to-dial feature or is there any other free softphone that supports? br Teppo

20 years, 6 months

[Serusers] SER and inter-domain routing

by Alexandru Deac

Hello, This might be a trivial issue, but I can't think of a solution to it. I have a SER installed on a linux machine acting as proxy for our VoIP network I'd like to configure so that it takes PSTN calls and forwards them to the proper GW for termination. However this is a 3rd paty GW over which we have no control. I've setup the ser to use rewritehostport("host", "port"); in order to send the call to the PSTN-GW but the call fails every time with a 404 Not found This i reported back by the PSTN GW. I belive this is due to the fact that the <To> field contains our realm but the remote GW cannot process it. Any idea how I can get ser to work with this? Has any one setup something similar? Any help would be appreciated, Alex.

20 years, 6 months

[Serusers] NAT & RTPPRoxy

by Howard March

Hi Gary, Did you get solution on how to change IP:port in the REGISTER packet to the extrenal NAT mappimg??? I'm also using RTC API 1.2. You help is apprecieated. Thankyou, BTW, anybody else has a solution for this?? Howard --------------------------------- Do you Yahoo!? Exclusive Video Premiere - Britney Spears

20 years, 6 months

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sr-users October 2003