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37453 discussions

[Serusers] ser 0.96 # mediaproxy # one-way audio SOLVED!!!
by Gareth Owen 08 Jan '07

08 Jan '07

Hi Davor I notice you had a problem with one way audio. I have the same problem but when the X-Lite client is behind firewall. Do you have any ideas or direction I could follow? Best regards Gareth Gareth Owen The information contained in this email message is intended only for the use of the individual(s) or entity to whom it is addressed, and may contain information that is confidential and may be legally privileged and exempt from disclosure under applicable laws. If the reader of this message is not the intended recipient or the employee or agent responsible to deliver it to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error please immediately notify the sender and destroy the original message from your system. The recipient should check this email and any attachments for the presence of viruses. BeVoice accepts no liability for any damage caused by any virus transmitted by this email

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[Serusers] Anyone like to finish my SER install for a fee?
by Bart Fisher 07 Jan '07

07 Jan '07

Hello all, I have SER 0.9.6 installed, along with Mediaproxy 1.8.0, Nathelper, SEMS 0.9.6, SerWeb 0.9.6 set up as a PSTN gateway on CentOS. However, there are some issues with NAT, Accounting (ACC) and the way I setup the web access. Also, I'd like to get to Voice mail and Conference working. Basically, I'd want you to take what's there and finish the install so everything works together. For an expert, all this is probably a few hours job. There is already SSH access in to the system. I started with a "LiveCD - VOIP" version of SER (http://www.wifi.com.ar/english/voip.html) with all the above (almost) but found it grossly out-dated and the Author unwilling to assist nor update to current releases. FYI: If you are are thinking of purchasing this software, save you money! The only thing nice is there is a prepaid billing script that might be converted to current releases. I just need to 'jump-start' this project to some usable level. I'm willing to pay for your time. If you are interested, please contact me at bart (at) icpage (dot) com. TIA Bart

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[Serusers] SER 0.9.6 and Media-Server Asterisk 1.4
by Stefan Tichy 07 Jan '07

07 Jan '07

Hi all, SER 0.9.6 is used as registrar/proxy and asterisk 1.4 as media server (voicemail, ...) The situation when phone A calls phone B and B is busy is handled by a failure route revert_uri(); rewritehostport("10.11.12.13:5070"); append_branch(); t_relay; This works fine, if for example phone A is some Snom 190. If another asterisk server is used instead of phone A it works if "pedantic=no" is set in the sip.conf of asterisk. If it is "pedantic=yes", then asterisk compares the tags in the To header fields. There is one tag from phone B and a second from the asterisk media server. Asterisk ignores every response not having the tag used by phone B. IMHO asterisk is wrong, but since I can't controll which asterisk servers are used as clients - is there a way to may asterisk happy? Would it be possible to remove tags from 1XX responses? -- Stefan Tichy <seruser(a)pi4tel.de>

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[Serusers] how many calls can SER support?
by zahra bahar 06 Jan '07

06 Jan '07

Hi. do you know how many calls can ser support? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Start SER with xlite
by zahra bahar 06 Jan '07

06 Jan '07

Hi. I installed SER as a proxy server and Xlite as a softphone.but xlite can't make a call. it sends :408 or 404 .is it any trouble with ser?or xlite?how can I find out ? what changes in ser.cfg are needed for having a pc to pc call with xlite? in ser.cfg: should fork=no or yes? is it necessary to use mysql module? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] proxy and redirect servers
by zahra bahar 06 Jan '07

06 Jan '07

Hi. Could any one say me when we use proxy server and when it is necessary to use redirect server? thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] TM : BYE retransmissions
by Jean-François SMIGIELSKI 05 Jan '07

05 Jan '07

Hello list! I observed, on an openser-1.1.0, that stateful relayings of BYE requests absorb retransmissions from the sender but do not retransmit it to the next hop in the Route header. Please, is it normal or is ti a misconfiguration from myself ? Thanks a lot! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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Re: [Serusers] aliases
by Nils Ohlmeier 05 Jan '07

05 Jan '07

As this might be interesting for other users I cc'ed the list again. Just to be sure: you know that their is no alias table any more, right? Alias or not is just a matter of the flag value in the uri table. In general there is no need any more for re-writting a request which was targeted towards an alias. Request towards an alias are handled exactly the same way as "normal" requests, because both will get the same UID value. If you used the alias table for re-distributing the requests outside of your domain, which is actually more a forwarding function then an alias, then you could have this function as one (out of many) possible alternative(s) with AVPs like this: (Note: non-tested pseudo code) lookup_user("$t.uid", "ruri"); ... if ($t.uid) { load_attrs("$tu", "$t.uid"); if ($t.forwarding_target) { attr2uri("$t.forwarding_target"); t_relay(); } } Greetings Nils On Friday 05 January 2007 17:27, you wrote: > Thanks for the reply, Nils! I suppose I am a more of a newbie than I first > appeared as your answer is a little over my head. What I am trying to do > with the new version of SER is what used to be done (in previous versions) > with > > If (lookup("aliases")){ > Route(1); > > To redirect users based on a few entries in the aliases table. I won't be > using the SER server as a registrar or to handle dynamic user locations or > even authenticated accounts -- it will be open to the public without > authentication to access SIP accessible numbers and users. Mostly what I > want to do is just to route calls that SER receives (like john(a)iptn.org, > 5555555555(a)iptn.org etc.) to the domain/uri that used to be accessed in the > aliases table (openser worked this way too). > > Sorry for not explaining better and thanks again for your help! > > Eliott > > > -----Original Message----- > From: Nils Ohlmeier [mailto:nils@iptel.org] > Sent: Friday, January 05, 2007 8:32 AM > To: serusers(a)lists.iptel.org > Cc: Eliott Spencer > Subject: Re: [Serusers] aliases > > Hello, > > On Friday 05 January 2007 04:27, Eliott Spencer wrote: > > I am coming over from OpenSER because I ran some tests and found SER to > > be much faster. There are some things that are new to me that I imagine > > are easy to do. I have a working config with the 12/22/06 development > > snapshot > > > of 0.10 but I would like to lookup aliases to redirect inbound calls to > > another domain. I filled in the domain table and the uri table with a > > user > > > and I looked through the sample config but can't seem to get it to work. > > I > > > have also loaded the domain module, then the uri module, then the avp and > > avpops modules (among others). > > > > > > > > Any idea what I might need to put in my route? This doesn't seem to work > > (stolen from ser.cfg.sample): > > > > > > > > if (lookup_user("$t.uid", "@ruri")) { > > > > setflag(1); > > > > route(1); > > > > break; > > > > }; > > from looking at your code snipset I'm not sure what you really want to > achieve. > In general you should just do the lookup_user and then examine the AVPs > afterwards. I think that it is easier then looking at the result code of > the > > lookup_user function. > If you really want to distingiush if the call was targeted to an alias or > the > non-alias URI of an user then check for the AVP ruri_canonical. If it is > set > > (to the value 1) the URI contained the canonical (=non-alias) URI of the > user. > If you just want to know if the URI is a know one and belongs to a user, > just > check for the presence of the uid AVP after the lookup_user call. If it is > present the user is know to the system under the value of the uid AVP. > Otherwise the user/URI is not known to the system. > > Hope this helps > Nils

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[Users] forking call to another user on same proxy
by Christian Schlatter 05 Jan '07

05 Jan '07

I try to fork a call to another user registered at the same proxy as the caller. I figured out that the easiest method would be to use the append_branch() function, something like route[0] { if (method=="REGISTER") { save("location"); exit; } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } # if someone calls test2@domain, fork call to test4@domain # (test2 and test4 are both registered at domain) if (avp_check("$rU", "eq/test2")) { append_branch("sip:test4@127.0.0.1"); } if (!t_relay()) { sl_reply_error(); }; exit; } I know that this could also be achieved by adding permanent location records, but I'd like to load the forking targets from a DB in the future. So my first question would be if using append_branch("sip:test4@127.0.0.1"); is the correct way to do parallel forking to a local user. I'm asking because this method does not work for me as expected. It seems that the two UAC state machines running on the same proxy host that are involved in a forking call do somehow not work well together. Forking the INVITE to both test2 and test4 works well, but when I pick up test2 the call gets terminated after a few seconds. Further investigations have revealed that canceling the second call leg does not work correctly. There is a message flow diagram available at http://www.unc.edu/~cschlatt/openser.html . For some reason, the proxy process sending from 127.0.0.1 is retransmitting the 487 "Request Terminated" although it receives ACKs from the upstream proxy. And this causes the upstream proxy to send a 487 back to the caller (time 15.360), which then terminates the call established between caller and test2. And this only happens if I fork to the same proxy, if I use two proxy hosts everything works fine (??). Any advice is highly appreciated, Christian

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[Users] integrating with Asterisk and OpenSER for Voicemail
by raviprakash sunkara 05 Jan '07

05 Jan '07

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport("192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060"); resetflag(2); } ......................... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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