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37555 discussions

[Users] Max-Forward header in Notify messages
by Toni Barata 16 May '07

16 May '07

Hi Anka, In Notify messages generated by Presence module the Max-Forward Header is missing. Some sip stacks have a strange behaviour when receiving Notify messages without Max-Forward header. Best regards, Toni

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[Users] purposely dropping reply message
by Tim 16 May '07

16 May '07

Hello, How/Where would be the best place to purposely drop a particular reply message ( > 400 ) I have armed both the reply route and failure route and attempted to drop the message by calling the drop() function but this doesn't seem to work. Is there a simple method for conditionally dropping a reply message? My goal is to force the UAC that created the original request to retransmit (UDP) the request in certain scenarios, such as when ACK to an initial invite has not been fully processed by the downstream UAS but a re-invite has been received by the UAS causing the UAS to return a 400 for the re-invite. I do not want the UAC to see the 400 reply, instead I simply want the UAC to resend the re-invite because no final response has been received, presumably by this time the ACK processing will be finished and the retransmitted re-invite would be handled smoothly. I realize replies are routed based on via headers. Is there some way to short circuit this routing other than drop()? Thanks Tim Tried this from failure route if(t_check_status("400")) { xlog( "L_ERR", "400 message in failure route 2 ********** \n" ); drop(); return; } Tried this from reply route if(status == "400") { xlog( "L_ERR", "400 in reply route 2 .... dropping\n" ); drop(); return; }

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[Users] trying to get 2-way audio in multi-homed openser configuration
by Taylor Carpenter 16 May '07

16 May '07

I have tried all sorts of configurations with both rtpproxy and media proxy, but I continue to end up with audio only working on one end of the call. The latest I tried to turn on rtpproxy for all calls. The call goes through but audio is only heard from the callee to the caller. Any help is greatly appreciated. Taylor

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[Users] XCAP Lite 1.0 released
by Adrian Georgescu 16 May '07

16 May '07

Change log xcap-lite (1.0.0) unstable; urgency=low * Added control interface for OpenSER to inform the Presence agent about changes in the presrules document. This allows the presence server to imediately send Notify to the updated buddy list after the end-user changes his presence policy document. Thanks to Anca Vamanu and Juha Heinanen for the support provided for adding this critical functionality * The configuration config.php has changed to allow for definition of the OpenSER management interface URL (refreshWatchersURL) * OpenSER must have the mi_xmlrpc module loaded and listening on the port configured in config.php * Fixed query in updateWatchers function * OpenSER trunk version 2230 or higher is required when refreshWatchersURL setting is enabled in config.php The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-1.0.0.tar.gz Regards, Adrian

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[Users] openser setup as a telephony gateway (to PSTN) for NAT'd clients
by Taylor Carpenter 16 May '07

16 May '07

I may be misunderstanding things (very probably so), but all the examples for both SER and OpenSER that I have seen either do not do NAT or NAT but are not the "end point" for a UAC to the PSTN. From what I can tell they are just sending on the SIP request to the next destination as is. The setup I am trying to accomplish is * SIP clients are all on a private network with connectivity to OpenSER directly on one interface (on the same private network) * OpenSER's other interface is on the external network (internet facing) * SIP clients are only sending telephone numbers (sip:telephone_number@*... do not know about PSTN providers) * OpenSER connects to the PSTN provider and send the number to dial that came from the SIP client * OpenSER "proxies" the entire call (with the help of rtpproxy or mediaproxy for RTP of course) * No incoming calls from the internet to OpenSER (no support for that is needed) * Registration not required for sip clients (they are all on same private network and authorized) I have found several posts, example configs, documents that have pieces of what I need (from what I can tell).. and I have tried to put it together, but it does not quite work... So is there some example that fits this type of usage? If not one then possibly several pieces from a few documents? I have been thinking that OpenSER setup as a outbound proxy configured for multihome, but everything I have seen on that just routes the calls through to where ever the SIP client was requesting as a final destination and I need to send to one destination (the PSTN provider). Any help is greatly appreciated. BTW, I was thinking the NAThelper or outbound proxy example from http://www.voip-info.org/wiki-SER+tips+and+tricks Looked close to what is needed, but have a bunch of stuff (seemingly) unneeded for my scenario and have nothing about PSTN connectivity. Thanks for taking the time to read this long post (if you made it this far). Taylor

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[Serusers] Race Condition on CANCEL?
by Gustavo Passos Tourinho 16 May '07

16 May '07

Hello, Im having some problems with cancelled calls. This is the scenario: U1 Proxy U2 INVITE -->>> <<--- 100 Trying INVITE -->>> <<--- 100 Trying <<--- 100 Trying CANCEL ->>> <<-- 200 Cancelling CANCEL ->> <<-- 180 Ringing <<-- 487 Cancelled <<-- 180 Ringing <<-- 200 OK (Wrong??) <<-- 200 OK My problem is that after some time waiting for "ringing", the user cancel the call. Even that proxy responses "487" it still forward the late 200 OK. Should it forward? I guess not because the transaction was destroyed, right? Can it be a configuration problem on my ser.cfg ou it can be in t_relay implementation? Thanks in advanced. Regards, Gustavo

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[Serusers] ser_fifo gives timeout
by Andrey Kuprianov 16 May '07

16 May '07

Hi guys, I dont know what the problem is.. Really.. Im out of ideas with permissions and reinstallations for serweb's CTD and IM. During my 1st installation with serweb they both worked fine. Then I had to reinstall my OS (Ubuntu 7.x btw) and now they give both timeouts.. I have even copied send_im() and write2fifo() functions separately into a PHP file and ran it from command line. Same old "408 Request Timeout".. Please, suggest something. Im basically using nat-mediaproxy.cfg for config file and I have tried setting various ownerships and permissions to serweb dir and for /tmp/serweb and /tmp/ser_fifo.. Nothing helps really. All packages are installed for php4 including php4-cli, pear, pear-log, pear-DB, XML_RPC and even classkit (which is useless here anyway).

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[Serusers] Configuration examples
by Nils Ohlmeier 15 May '07

15 May '07

Hello everybody, as we want to provide more examples for the configuration the following the question came up: Do you prefer - one big example configuration - probably harder to understand - easier to maintain for us - one example config file per "feature" - probably easier to understand your desired "feature" - you might have problems to integrate all your desired "features" into one config without breaking anything - harder to maintain for us, as we have to update several files :-) As the configuration examples are indented to be for the community, I would like to know what you prefer or if you see other options. Thanks Nils

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[Users] SIP SIMPLE / XCAP test bed - feedback request
by Adrian Georgescu 15 May '07

15 May '07

Hello, We setup a SIP SIMPLE testbed using the latest OpenSER / XCAP-lite. To make use of this testbed you need a recent Eyebeam commercial version for Apple or Windows. Other SIP/SIMPLE/XCAP client implementations are of course more then welcome and desired. To use this test bed register a SIP account at: http://sip2sip.info Instructions about the SIP/Presence/XCAP settings are sent by email to the registrant. We appreciate any feedback regarding the software behavior so that we can fix any bugs in the used implementations and update the wiki pages with proper configuration samples. Thank you for your support, Adrian

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Re: [Serusers] SER/PSTN Gateway conf
by flavio 15 May '07

15 May '07

My Record_Route and Loose section are the follow: # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; As follow reported, SER Record Route INVITE original INVITE message, that lack of 'Route:' header, and forward it to IP Phone. Why not for BYE? =_=" Thanks for support and big patience. ^_^ U 2007/05/15 16:14:20.789400 10.28.52.105:5060 -> 10.28.19.202:5060 INVITE sip:06605XXXXX@10.28.19.202 SIP/2.0. To: <sip:06605XXXXX@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Remote-Party-ID: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off. Contact: <sip:10.28.52.105:5060>. Call-ID: 238757681179426(a)10.28.52.105. Max-Forwards: 70. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY. CSeq: 23392 INVITE. Content-Length: 274. Content-Type: application/sdp. . v=0. o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105. s=SIP Call. c=IN IP4 10.28.52.105. t=0 0. m=audio 16572 RTP/AVP 18 8 0. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 2007/05/15 16:14:20.790166 10.28.19.202:5060 -> 10.28.52.105:5060 SIP/2.0 100 trying -- your call is important to us. To: <sip:06605XXXXX@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Call-ID: 238757681179426(a)10.28.52.105. CSeq: 23392 INVITE. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=3803 req_src_ip=10.28.52.105 req_src_port=5060 in_uri=sip:0660522014@10.28.19.202 out_uri=sip:06605XXXXX@10.28.19.124;user=phone via_cnt==1". . # U 2007/05/15 16:14:20.790257 10.28.19.202:5060 -> 10.28.19.124:5060 INVITE sip:06605XXXXX@10.28.19.124;user=phone SIP/2.0. Record-Route: <sip:10.28.19.202;ftag=DXsf22387576811Yaz079427;lr=on>. To: <sip:0660522014@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.19.202;branch=z9hG4bK4639.ea946db1.0. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Remote-Party-ID: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off. Contact: <sip:10.28.52.105:5060>. Call-ID: 238757681179426(a)10.28.52.105. Max-Forwards: 16. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY. CSeq: 23392 INVITE. Content-Length: 274. Content-Type: application/sdp. . v=0. o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105. s=SIP Call. c=IN IP4 10.28.52.105. t=0 0. m=audio 16572 RTP/AVP 18 8 0. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. 2007/5/15, Kostas Marneris <K.Marneris(a)otenet.gr>: - Nascondi testo tra virgolette - > As far as I know and understand it seems that the > R-URI of the BYE mesg (1st line) which GW sends to SER is wrong. > > I expected to see : > BYE sip:user@SIP_Phone_IP_Address SIP/2.0 > > or the problem is that the 'Route:' header is missing from the BYE mesg > so it's not loose_routed. > > Do you Record_Route the original INVITE ? > > Check also : rfc3261 / 16.12.1 > > > Kostas > >

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