sr-users October 2006

sr-users@lists.kamailio.org

189 participants
345 discussions

[Serusers] rtp-asymmetric-clients problem.
by Ricardo Martinez 18 Oct '06

18 Oct '06

Hello User. I using a rtp-asymmetric client in my platform, i want to add this client as a asymmetric client so i using this configuration. [ser.cfg] modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients") [rtp-asymmetric-clients file in /usr/local/etc/ser/] # # Lines starting with a '#' character or empty lines are ignored. # Put a User-Agent name or regular expression per line. # Check is case insensitive. # # This file should only list SIP clients that are asymmetric regarding # the RTP media streams. For clients that are asymmetric regarding the SIP # signaling, use the equivalent SIP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # SIPquest-SIPH323-Gateway/R2-23 When the 200 - OK from the asymmetric client arrive : SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.148.246;branch=z9hG4bK9eda.3d49eaf1.0. Via: SIP/2.0/UDP 192.168.1.100;received=100.100.148.248;rport=46107;branch=z9hG4bKc0a801640000004b453532800000783900000087. Record-Route: <sip:100.100.154.36;ftag=10302336931863;lr=on>. Record-Route: <sip:7072408196@100.100.148.246:5060;nat=yes;ftag=10302336931863;lr=on>. From: "unknown"<sip:5502408196@sip.desa.mydomain.net>;tag=10302336931863. To: <sip:7072408196@sip.desa.mydomain.net>;tag=3235014976. Call-ID: E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100. Cseq: 2 INVITE. Date: Tue, 17 Oct 2006 19:44:02 GMT. Server: SIPquest-SIPH323-Gateway/R2-23. Content-Type: application/sdp. Content-Length: 162. Contact: sip:7072408196@100.100.154.119. . v=0. o=100.100.154.119 1 627067186 IN IP4 100.100.154.119. s=SIP Library call. c=IN IP4 100.100.148.230. t=3370103042 0. m=audio 17322 RTP/AVP 0. a=rtpmap:0 PCMU/8000. Oct 17 16:45:18 sip mediaproxy[1783]: command lookup E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100 100.100.148.230:17322:audio 100.100.154.36 sip.desa.mydomain.net local sip.desa.mydomain.net unknown SIPquest-SIPH323-Gateway/R2-23 info=from:5502408196@sip.desa.mydomain.net,to:7072408196@sip.desa.mydomain.net,fromtag:10302336931863,totag:3235014976 Oct 17 16:45:18 sip mediaproxy[1783]: command execution time: 0.55 ms Oct 17 16:45:18 sip mediaproxy[1783]: session E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100: caller signed in from 100.100.148.248:49172 (RTP) (will return to 100.100.148.248:49172) Oct 17 16:45:18 sip mediaproxy[1783]: session E2BBA13E-A3CA-4E54-91C0-C30AF6466324(a)192.168.1.100: called signed in from 100.100.148.230:17090 (RTP) (will return to 100.100.148.230:17090) There is no match with the rtp-asymmetric-clients file ? What i'm doing wrong? I hope that someone can help me here. Thanks Ricardo.-

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[Serusers] Re: [Serdev] pa module: presence "substatus"
by Vaclav Kubart 18 Oct '06

18 Oct '06

Hi, notes are quite often used for this. Can you send your config here? What SER version do you use? And if you try register your users via sending REGISTER messages it is still the same? Vaclav P.S. Send these messages to serusers too, might be that somebody else will have the same problem... On Tue, Oct 17, 2006 at 05:18:55PM +0200, Diego Do?ate wrote: > > Hi, > > Thanks, then I will use a <note> tag for the presence > substatus(<note>Away<\note>, for instance). > > I have a problem with the Subscription. User A subscribes to the presence > of user B and receives a NOTIFY but with "close/offline", and both users are > previously registered successfully in SER. > > Both users are added with serctl, but it seems the SER does not "save" > user A registration when user B subscribes... Where may be the problem? May > it be related to autherization? By now, it is disabled... > > Thanks in advance > > > > -----Mensaje original----- > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Enviado el: lunes, 16 de octubre de 2006 18:57 > Para: Diego Do?ate > CC: serdev(a)lists.iptel.org > Asunto: Re: [Serdev] pa module: presence "substatus" > > Hi, > support for PIDF extensions (elements from non-pidf namespace) is nearly > finished in current CVS version, but it still needs some work. It will be > done as soon as possible. > > Notes should be possible in last presence snapshot without any problems > - they can be within tuple or presence elements according specification. > > Vaclav > > On Mon, Oct 16, 2006 at 05:52:47PM +0200, Diego Do?ate wrote: > > > > Hi, > > > > In the SER "pa" module, I would like to use a tag like "note" or > > "substatus" in the pidf of the NOTIFY when a PUBLISH is received (with > > a presence change, to "Away", for instance), but SER only modified the > > <status> tag ("open"/"closed"). > > > > How can I implement a more detailed presence status management in > > the SER? > > > > Thanks in advance > > > > > > _______________________________________________ > > Serdev mailing list > > Serdev(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serdev >

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[Serusers] serctl ping 408
by Ali Zaidi 18 Oct '06

18 Oct '06

Hi, Just wondering if someone is facing similar problem. My setup is SER/MediaProxy/MYSQL on public IP, I have couple of dozen SPA-1001 all over the internet. The problem I'm facing is that many of these ATA SPA-1001 some times do not ring, so try to ping (serctl ping sip:550505050@myserver.com) and i get error 408. When i restart these ATA everthing starts working but couple of hours later, it's the same problem. Not to mention that (serctl showdb) shows that sip client is registered, however I still get 408. I tried the following: (but no success) 1. SPA-1001 change Register Expires: from 3600 to 600 2. re-INVITE TO 30 Can someone help, or maybe there are some manufacturing bug on some linksys SPA-1001, by the way all my ATA use uniform configs and firmware. If someone is willing to look into the ngrap, i can show it to you. Thanks, Ali...

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[Users] presence agent?
by Cesc 17 Oct '06

17 Oct '06

Hi, I saw some chatter a few days ago about openser's presence agent being about to be put on the CVS ... any idea on the ETA? Two students are coming to realize a project for a SIP-based content routing module, and the PA could be a good base. They should be using either ser's pa or openser's pa for processing the sub/not/pub messages, probably extending a bit the functionality ... And if the PA is already available (say ... 2 weeks), i can force it on them to use. Regards, Cesc

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[Users] Syslog tutorial
by Marnus van Niekerk 17 Oct '06

17 Oct '06

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[Users] calls being disconnected
by Richard Bennett 17 Oct '06

17 Oct '06

Hi, I am having increasing problems with calls being cut-off after a few seconds. I'm running openser with this version: version: openser 1.1.0-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ main.c compiled on 04:14:03 Aug 11 2006 with gcc 4.1.1 with mediaproxy 1.7.2 running on the same dual cpu server. At the time of the disconnect there were 100 calls going over mediaproxy (cpu at 46%), but it also happens with lower load. The pstn gateway is a Lucent Max TNT. My config is based on the onsip gettingstarted examples, except that all calls are sent over mediaproxy, by setting use_media_proxy even if no nat problems are detected: if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); ## use_media_proxy(); }; use_media_proxy(); #use mediaproxy always and: #if (isflagset(6) || isflagset(7)) { # if (!isflagset(8)) { # setflag(8); # use_media_proxy(); # }; #}; if (!isflagset(8)) { setflag(8); use_media_proxy(); }; I can attach the full config if required. I had one problem when upgrading to openserv1.1,0, that this didn't work anymore: avp_write("i:45", "inv_timeout"); in the pstn route. So this is missing, and the module section has been changed: modparam("tm", "fr_inv_timer", 27) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") I tried finding the correct syntax from the latest documentation, but nothing worked. Could this be causing call disconnects? I attached a ngrep trace of a call that disconnected after a few seconds. It looks like my ua sends two ACKs to the gateway, doesn't receive a reply, and then send a bye, but I'm not sure why all that happens. Any help or pointers is appreciated. Richard

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[Users] A question about acc flags..
by nick 17 Oct '06

17 Oct '06

I'm doing dfferent levels of accounting, radius for billing concerns, mysql for some further in depth analysis. What I'm wondering about is the numbers used in the modparam() lines, I know that the params for radius_flag, radius_missed_flag, etc need to be different, but can the numbers for radius_flag, db_flag and log_flag overlap? for example, all _flag acc would be one, all _missed_flag 2, etc etc?? I haven't seen anything that addressed the issue, either as a possibility or as something to avoid, do I need to worry about it? Nick

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[Users] openserctl and permanent ul contacts
by Andreas Granig 17 Oct '06

17 Oct '06

Hi, I've some registrars and proxies using db_mode 3 on OpenSER 1.1.0. When I try to add a permanent contact into the location table using openserctl, the "permanent"-flag (128) is set, but the contact is removed upon hit of next timer. I quickly checked the source code, but couldn't find any timer handling for mode DB_ONLY, so maybe one of the core developers could take a look. Thanks, Andy

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[Users] REGISTER problems
by Jon Schøpzinsky 17 Oct '06

17 Oct '06

Hello Im having a somewhat weird problem, at least for somebody like me, who is accustomed to the lovely NAT handling functionality of Asterisk. When my phone registers on our internal network, everything works like a charm, and the user is authenticated. But when I move the phone out on the public internet, it doesn't. I can see that the only thing changed in the sip messages, is our Cisco PIX firewall, that changes a few of the IP addresses. How can this have an impact on the calculation of the HA1 string, since this is where it fails. The realm hasn't changed, so I cant understand why this doesn't work. Could somebody who is a little more accustomed to OpenSER tell me why this happends? Med venlig hilsen Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefoni Tlf.: +45 70 20 40 59 Direkte: +45 88 20 03 36 Mobil: +45 31 20 67 09 http://www.detele.dk Dansk Erhvervs-Telefon er den mest fleksible og service-orienterede teleudbyder til dansk erhvervsliv. Derfor leverer vi til stadighed nye innovative og værdiskabende telefoniløsninger til danske virksomheder. Produkterne bygger på vores grundlæggende filosofi om, at overblik og service skal gå hånd i hånd med den rigtige pris. Hos Dansk Erhvervs-Telefoni, mener vi, teknologivalget skal styres af behovene hos den enkelte virksomhed. Vi tilbyder såvel løsninger baseret på Voice over IP (VoIP) som løsninger baseret på traditionel ISDN-teknologi. Vi mestrer teknologierne, men vores drivkraft er kundens behov. Det er vigtigst for os, at rådgive ud fra en behovsanalyse - ikke ud fra teknologifascination. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.4/476 - Release Date: 14-10-2006

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[Users] Flag
by Gregoire 17 Oct '06

17 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! Does anyone have a list of the 32 flags that can be set in the configuration, and their signification? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFM4nHI8gmGeMTr0sRAuKMAJ9Y32PNwMyQwdYi+REYX30wnnl91ACdFhrg /DZnyUUakReZTObMgQuKU50= =TTZC -----END PGP SIGNATURE-----

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sr-users October 2006