- sr-users - kamailio.org

[Users] avp_db_load() generates core dump
by Christian Schlatter 13 Apr '06

13 Apr '06

Using the (g)lobal flag together with avp_db_load() crashes my OpenSER in a reproducible manner. The config works if I don't use the g flag and I tested it on CVS current. I try to give as much information as possible here: from openser.cfg: ... # check for Call Forwarding Unconditional (CFU) if (avp_db_load("$ruri/username", "$avp(s:cfu)")) { avp_print(); # replace R-URI with CFU URI, g will fetch all CFU URIs --> request will fork avp_pushto("$ru", "$avp(s:cfu)/g"); # avp_pushto("$ru", "$avp(s:cfu)"); --> this works sl_send_reply("181", "Call is being forwarded (CFU)"); route(ROUTE_DEFAULT_RELAY); exit; } ... from openser.log: ... openser[31411]: INFO:avpops:print_avp: p=0xb5770dc8, flags=103 openser[31411]: INFO: name=<cfu> openser[31411]: INFO: val_str=<sip:vm-admin@xx.xx.xx.xx / 25> openser[31411]: INFO:avpops:print_avp: p=0xb5770d98, flags=2 openser[31411]: INFO: id=<34> openser[31411]: INFO: val_str=<sip:cschlatt@unc.edu / 20> openser[31403]: child process 31411 exited by a signal 11 ... from core dump: #0 ops_pushto_avp (msg=0x8123bb0, dst=0x8123130, src=0x8123160) at avpops_impl.c:1111 1111 flags = avp->flags; Thanks Christian

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[Users] Include Function?
by Tavis P 13 Apr '06

13 Apr '06

Has anyone though about creating an include() core function in OpenSER? I've got several proxies running and the only real difference between the OpenSER scripts on them is the database parameters and alias definitions so if there was an include function i wouldn't have to edit the the script each time i make an update on one system that i need to propagate to all the rest, i'd just have to update the main "script" file and it would simply include another file that would continue the unique parameters In retrospect this sounds a little self serving but it would be a benefit to all i'm sure! =D

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RE: [Serusers] Problem in SERWEB configuration
by Gould, Aaron 13 Apr '06

13 Apr '06

by the way, i'm running ser on solaris 10. is that ok for this serweb and db you suggest. i donwloaded "serweb-0.9.4.tar.gz" is this the correct serweb file i downloaded "DB-1[1].7.6.tgz" is this the correct DB file to install thanks aaron ________________________________ From: Jahangir Mushtaq [mailto:jahangir19m72@yahoo.com] Sent: Thu 4/13/2006 8:46 AM To: Gould, Aaron Subject: RE: [Serusers] Problem in SERWEB configuration hi aaron Sure.. here is the link http://www.iptel.org/ser/serweb/ file name is "serweb" you also need to download and install "DB" http://www.pear.php.net/package/DB regard Jahangir --- "Gould, Aaron" <aaron.gould(a)ngc.com> wrote: > Jahangir, sorry i won't be able to help you at this > point. i'm new to ser also. but maybe you can help > me. what did you download for serweb? please send > specific file names and links. > > thanks > aaron > > ________________________________ > > From: serusers-bounces(a)lists.iptel.org on behalf of > Jahangir Mushtaq > Sent: Thu 4/13/2006 3:56 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Problem in SERWEB configuration > > > > Hi everybody > Need help from some one.New in SER > Configuration...... > I did configur SER and now want to configure > SERWEB.i > download and install it. i access "index.php" in > browser and it states username and password.I > subscribe user name and password but it gives the > message Bad usersname or password. > > what would be the problem.? > > regards > > Jahangir > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] log from rtpproxy, send a 0 as voice port
by laurent schweizer 13 Apr '06

13 Apr '06

hello, this is the log from the rtpproxy and i see that the returned port is 0. caller's address filled in: 83.76.253.30:55351 (RTP) callee's address filled in: 193.zzzzzzzzz:19320 (RTP) guessing RTCP port for callee to be 19321 received command "U 3420006225(a)192.168.1.33 193.zzzzzzzz 19320 as6e4b6526;1 3835457540" adding strong flag to existing session, new=1/0/0 lookup on ports 35008/0, session timer restarted pre-filling callee's address with 193.zzzzzzzzz:19320 sending reply "0 62.2.xxxxxx -------------------- all logs received command "U 3420006225(a)192.168.1.33 83.76.253.30 10600 3835457540;1" new session 3420006225(a)192.168.1.33, tag 3835457540;1 requested, type strong new session on a port 35006 created, tag 3835457540;1 pre-filling caller's address with 83.76.253.30:10600 sending reply "35006 62.2.xxxxxxxx " received command "U 3420006225(a)192.168.1.33 83.76.253.30 10700 3835457540;2" new session 3420006225(a)192.168.1.33, tag 3835457540;2 requested, type strong new session on a port 35008 created, tag 3835457540;2 pre-filling caller's address with 83.76.253.30:10700 sending reply "35008 62.2.xxxxxxx " received command "L 3420006225(a)192.168.1.33 62.2.yyyyyy 13908 3835457540;1 as6e4b6526" lookup on ports 35006/35010, session timer restarted pre-filling callee's address with 62.2.195.189:13908 sending reply "35010 62.2.xxxxxxxx " caller's address filled in: 83.76.253.30:55351 (RTP) callee's address filled in: 193.zzzzzzzzz:19320 (RTP) guessing RTCP port for callee to be 19321 received command "U 3420006225(a)192.168.1.33 193.zzzzzzzz 19320 as6e4b6526;1 3835457540" adding strong flag to existing session, new=1/0/0 lookup on ports 35008/0, session timer restarted pre-filling callee's address with 193.zzzzzzzzz:19320 ---------- >>>>>>>>>> >>>> sending reply "0 62.2.xxxxxx " received command "L 3420006225(a)192.168.1.33 192.168.1.33 10600 as6e4b6526;1 3835457540" lookup on ports 35012/0, session timer restarted pre-filling caller's address with 192.168.1.33:10600 sending reply "35012 62.2.xxxxxxx " received command "D 3420006225(a)192.168.1.33 3835457540 as6e4b6526" forcefully deleting session 2 on ports 35012/0 RTP stats: 308 in from callee, 0 in from caller, 308 relayed, 0 dropped RTCP stats: 3 in from callee, 0 in from caller, 3 relayed, 0 dropped session on ports 35012/0 is cleaned up forcefully deleting session 1 on ports 35006/35010 RTP stats: 267 in from callee, 687 in from caller, 954 relayed, 0 dropped RTCP stats: 1 in from callee, 0 in from caller, 1 relayed, 0 dropped session on ports 35006/35010 is cleaned up sending reply "0

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[Users] Media proxy
by Rafael Neves Harff 13 Apr '06

13 Apr '06

Hi all! I've got openser + mediaproxy running. I have 2 questions: 1) I'm doing a PESQ comparation betwin Asterisk and Openser + mediaproxy. I tested both software in the same lab and I note that media proxy route RTP packets more slowly than Asterisk. Is it true? Can I change any seeting to improove mediaproxy speed/quality? 2) I need to test it with differend CODEC's. I'm using Express Talk as UA to record the call to compare using PESQ software, but using this softphone I ca't set which CODEC to use. Can I set which CODEC is allowed in Openser/Mediaproxy software? Is there a UA that can play/reccord .WAV files which allow to choose the CODEC? Thanks once again. Rafael

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[Users] minor doc issue with mediaproxy module
by Christian Schlatter 13 Apr '06

13 Apr '06

It seems that the mediaproxy module depends on the domain module (mediaproxy uses the is_from_local() and is_uri_host_local() functions). I think this should be added to the "Dependencies" section of mediaproxy. Interestingly, the mediaproxy works fine for me even without any actual data in the domain SQL table. I also don't see why the mediaproxy needs to know if a call is from local or not.... I'd appreciate if someone could enlighten me ;-) Christian

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[Serusers] ERROR message : MySQL server has gone away
by "한승전" 13 Apr '06

13 Apr '06

please help..... If i restart MySQL(version 5.0.19), SER shows next error message. Mar 31 22:37:18 localhost /usr/local/sbin/ser[32701]: submit_query(): MySQL server has gone away Mar 31 22:37:18 localhost /usr/local/sbin/ser[32701]: submit_query(): Error while submitting query Mar 31 22:37:18 localhost /usr/local/sbin/ser[32701]: get_ha1(): Error while querying database Mar 31 22:37:22 localhost /usr/local/sbin/ser[32703]: Binding '1234(a)192.9.2.1','sip:1234@192.9.2.11:5060' has expired As appear in upside message, SER does not auto-reconnect to mysql. If MySQL restarted, SER must be able to do Auto-Reconnect. Is no there way that SER can do Auto-Reconnect to MySQL?

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[Serusers] force_rtp_proxy2: incorrect port in reply
by Laurent Schweizer 13 Apr '06

13 Apr '06

Hello, I have this error with the nathelper and rtpproxy : force_rtp_proxy2: incorrect port in reply from rtp proxy. How can I have log from the rtpproxy ? Laurent

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RE: [Serusers] Problem in SERWEB configuration
by Gould, Aaron 13 Apr '06

13 Apr '06

Jahangir, sorry i won't be able to help you at this point. i'm new to ser also. but maybe you can help me. what did you download for serweb? please send specific file names and links. thanks aaron ________________________________ From: serusers-bounces(a)lists.iptel.org on behalf of Jahangir Mushtaq Sent: Thu 4/13/2006 3:56 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Problem in SERWEB configuration Hi everybody Need help from some one.New in SER Configuration...... I did configur SER and now want to configure SERWEB.i download and install it. i access "index.php" in browser and it states username and password.I subscribe user name and password but it gives the message Bad usersname or password. what would be the problem.? regards Jahangir __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] force_rtp_proxy2: incorrect port in reply
by Laurent Schweizer 13 Apr '06

13 Apr '06

Hello, I have this error with the nathelper and rtpproxy : force_rtp_proxy2: incorrect port in reply from rtp proxy. How can I have log from the rtpproxy ? Laurent

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[Users] t_relay()
by Simon Morvan 13 Apr '06

13 Apr '06

Hello there, Just to be sure : if I t_relay() an INVITE to a remote AS, will OpenSER automatically discard re-emmitted INVITES from UAC ? -- Simon.

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Re: [Users] Allow only TLS connections
by Daniel-Constantin Mierla 13 Apr '06

13 Apr '06

Hello, maybe the clients register non-TLS contacts, take a look in the location table. Also, in aliases, you may have some addresses that point to external domains. Cheers, Daniel On 04/13/06 12:05, Christoph Fürstaller wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi Daniel, > > Daniel-Constantin Mierla wrote: > >> Hello, >> >> On 04/13/06 11:52, Christoph Fürstaller wrote: >> >> Hi, >> >> I tried that out. I check if proto is TLS: >> if (proto != TLS) { >> sl_send_reply("403", "Forbidden"); >> exit; >> }; >> >> But I get this error: >> 3(28893) ERROR:tm:add_uac: can't fwd to af 2, proto 1 (no >> corresponding listening socket) >> 3(28893) ERROR:tm:t_forward_nonack: failure to add branches >> 3(28893) ERROR:tm:t_relay_to: t_forward_nonack returned error >> >> What does it mean? What I'm doing wrong? >> My SER is only listening on tls port 5061. Do I still have to open udp >> 5060 ? >> >> >> >>> it seems that you try to forward on UDP. >>> > I figured that out too. But I don't know which part forwardes something > on UDP? I attached my conf. Can you give it a quick look? > > >>> You can configure openser to >>> listen on UDP as well, and drop messages coming on UDP, if you want to >>> accept only TLS. (as you have in above snippet). If all peers you >>> connect to support TLS, then you can forse sending over TLS all the time. >>> >>> Cheers, >>> Daniel >>> > > chris... > >> Cesc wrote: >> >> >> >>>>> http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook&DokuWiki=6c17… >>>>> >>>>> >>>>> >>>>> >>>>> On 4/11/06, Thorsten.Haupt(a)t-systems.com >>>>> <Thorsten.Haupt(a)t-systems.com> wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> I searched for this function, but I didn't found it :-( >>>>>> Knows anyone the correct code, not only pseudo-code? >>>>>> >>>>>> Torsten >>>>>> >>>>>> -----Ursprüngliche Nachricht----- >>>>>> Von: Cesc [mailto:cesc.santa@gmail.com] >>>>>> Gesendet: Dienstag, 11. April 2006 14:03 >>>>>> An: Haupt, Thorsten >>>>>> Cc: users(a)openser.org >>>>>> Betreff: Re: [Users] Allow only TLS connections >>>>>> >>>>>> I think in openser there is a function to check what transport the >>>>>> message came in ... you can do something like: >>>>>> if ( transport != TLS ) { >>>>>> send error to UA >>>>>> break; >>>>>> } >>>>>> >>>>>> Cesc >>>>>> >>>>>> On 4/11/06, Thorsten.Haupt(a)t-systems.com >>>>>> <Thorsten.Haupt(a)t-systems.com> wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I use OpenSER in a testing environment for VoIP security. My clients >>>>>>> connect via TLS. If I deactivate UDP/5060 on the server, it doesn't >>>>>>> work correct. >>>>>>> Some Clients can't connect and others can't establish calls. I read in >>>>>>> another thread, that UDP is mandatory for SIP and that the server >>>>>>> need it. >>>>>>> >>>>>>> But how can I prevent users from connecting via UDP and force them to >>>>>>> use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But is >>>>>>> this the correct way? Are there any parameters server-side to force >>>>>>> users to connect via TLS? >>>>>>> >>>>>>> Thanks for response. >>>>>>> Torsten >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users(a)openser.org >>>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)openser.org >>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.1 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org > > iD8DBQFEPhRDR0exH8dhr/YRAkatAJ9Y1nNS7h3Y/TyrvxnPIgBUGe8UfwCcCBwZ > grA83KOBRq5hOaEoK8mbaY4= > =YrlR > -----END PGP SIGNATURE----- > >

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[Serusers] Serusers Digest, Vol 36, Issue 13
by cjordan＠econetwireless.com 13 Apr '06

13 Apr '06

THIS IS AN AUTOMATED RESPONSE Our offices are closed from 16h00 on WEDNDESDAY, 12 April 2006 until Tuesday, 18 April 2006. Your email has not been read. .. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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[Users] Gateway disconnects call after about 10 seconds
by Cameron Beattie 13 Apr '06

13 Apr '06

I am forwarding calls to a PSTN gateway with authentication provided by the UAC module. The call connects and two-way audio is heard. Then, after about 10 seconds, the call is disconnected. If I register a UAC directly with the gateway the call continues until either party hangs up but when openSER is in the path (just the signalling, not the audio) the call gets disconnected. The gateway is reported as Cisco-SIPGateway/IOS-12.x but SER is in front of that i.e. (Grandstream<->openSER)<->(SER<->Cisco). The SIP traffic from the INVITE until the call disconnects is as follows. Note that no SIP traffic is seen at the openSER proxy at the point of disconnect and the disconnect happens a few seconds after the final ACK: U 60.234.100.100:5060 -> 147.202.100.100:5060 INVITE sip:006495551122@beta.domain.co.nz SIP/2.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz> Contact: <sip:6495551122@60.234.100.100> Supported: replaces Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 360 v=0 o=6495551122 8000 8000 IN IP4 60.234.100.100 s=SIP Call c=IN IP4 60.234.100.100 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 #U 147.202.100.100:5060 -> 60.234.100.100:5060SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 147.202.100.100:5060 "Noisy feedback tells:pid=4622 req_src_ip=60.234.100.100 req_src_port=5060 in_uri=sip:006495551122@beta.domain.co.nz out_uri=sip:006495551122@203.184.100.100 via_cnt==1" ##U 147.202.100.100:5060 -> 203.184.100.100:5060 INVITE sip:006495551122@203.184.100.100 SIP/2.0 Record-Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz> Contact: <sip:6495551122@60.234.100.100> Supported: replaces Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 360 v=0 o=6495551122 8000 8000 IN IP4 60.234.100.100 s=SIP Call c=IN IP4 60.234.100.100 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 #### U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=df74dfb1855fc2bffa5454d042443f01.b99c Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 INVITE Proxy-Authenticate: Digest realm="gateway.co.nz", nonce="443b3239652d054daebf907b011b989f20f71fd5" Server: Sip EXpress router (0.8.14-6 (i386/linux)) Content-Length: 0 Warning: 392 203.184.100.100:5060 "Noisy feedback tells:pid=18054 req_src_ip=147.202.100.100 req_src_port=5060 in_uri=sip:006495551122@203.184.100.100 out_uri=sip:006495551122@gateway.co.nz:5060 via_cnt==2" #U 147.202.100.100:5060 -> 203.184.100.100:5060 ACK sip:006495551122@203.184.100.100 SIP/2.0 Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.0 From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 To: <sip:006495551122@beta.domain.co.nz>;tag=df74dfb1855fc2bffa5454d042443f01.b99c CSeq: 52471 ACK User-Agent: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 #U 147.202.100.100:5060 -> 203.184.100.100:5060 INVITE sip:006495551122@203.184.100.100 SIP/2.0 Record-Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz> Contact: <sip:6495551122@60.234.100.100> Supported: replaces Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 360 Proxy-Authorization: Digest username="645566778", realm="gateway.co.nz", nonce="443b3239652d054daebf907b011b989f20f71fd5", uri="sip:006495551122@203.184.100.100", response="8a49e7678e33108ac0771c7eaf60b7ad", algorithm="MD5" v=0 o=6495551122 8000 8000 IN IP4 60.234.100.100 s=SIP Call c=IN IP4 60.234.100.100 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 #####U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 INVITE Server: Sip EXpress router (0.8.14-6 (i386/linux)) Content-Length: 0 Warning: 392 203.184.100.100:5060 "Noisy feedback tells:pid=18064 req_src_ip=147.202.100.100 req_src_port=5060 in_uri=sip:006495551122@203.184.100.100 out_uri=sip:8888095551122@202.180.100.100:5060 via_cnt==2" ########U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1,SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes #U 147.202.100.100:5060 -> 60.234.100.100:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes ############################################# U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1,SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes #U 147.202.100.100:5060 -> 60.234.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes ##### U 60.234.100.100:5060 -> 147.202.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK0f1649effc02b74c Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 #U 147.202.100.100:5060 -> 203.184.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Record-Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Via: SIP/2.0/UDP 147.202.100.100;branch=0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK0f1649effc02b74c Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 ####U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1,SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes #U 147.202.100.100:5060 -> 60.234.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes ####U 60.234.100.100:5060 -> 147.202.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK3ef8f33855e9ca1f Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 #U 147.202.100.100:5060 -> 203.184.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Record-Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Via: SIP/2.0/UDP 147.202.100.100;branch=0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK3ef8f33855e9ca1f Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 ####### U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1,SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes #U 147.202.100.100:5060 -> 60.234.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes ####U 60.234.100.100:5060 -> 147.202.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK66362c003682c65c Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 #U 147.202.100.100:5060 -> 203.184.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Record-Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Via: SIP/2.0/UDP 147.202.100.100;branch=0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK66362c003682c65c Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 ############## U 203.184.100.100:5060 -> 147.202.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.100.100;branch=z9hG4bK5493.591f2534.1,SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes #U 147.202.100.100:5060 -> 60.234.100.100:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bK2936917715c397f4 From: <sip:645566778@gateway.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Date: Tue, 11 Apr 2006 04:31:09 GMT Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 Server: Cisco-SIPGateway/IOS-12.x CSeq: 52471 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:8888095551122@202.180.100.100:5060> Record-Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on>,<sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 8327 3527 IN IP4 202.180.100.100 s=SIP Call c=IN IP4 203.184.100.100 t=0 0 m=audio 35366 RTP/AVP 0 c=IN IP4 203.184.100.100 a=rtpmap:0 PCMU/8000 a=ptime:20 a=nortpproxy:yes #####U 60.234.100.100:5060 -> 147.202.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bKf4eefea8493b1705 Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 #U 147.202.100.100:5060 -> 203.184.100.100:5060 ACK sip:8888095551122@202.180.100.100:5060 SIP/2.0 Record-Route: <sip:147.202.100.100;ftag=17f4875c4046b39d;lr=on> Via: SIP/2.0/UDP 147.202.100.100;branch=0 Via: SIP/2.0/UDP 60.234.100.100;branch=z9hG4bKf4eefea8493b1705 Route: <sip:006495551122@203.184.100.100;ftag=17f4875c4046b39d;lr=on> From: <sip:6495551122@beta.domain.co.nz>;tag=17f4875c4046b39d To: <sip:006495551122@beta.domain.co.nz>;tag=3DD382F8-2296 Contact: <sip:6495551122@60.234.100.100> Call-ID: 90fb36fc2ad0ad5a(a)192.168.0.11 CSeq: 52471 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Relevant parts of openSER.cfg: route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, #f (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("147.202.100.100:5060;nat=yes"); # insert IP address #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); #call end_media_session in case we're proxying media end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= if (has_totag() && (method=="INVITE" || method=="ACK")) { #check client is nated or that we've already identified it's nated if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); return; }; if (uri==myself) { if (method=="ACK") { route(6); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); log(1,"route[2] (Register handler"); #Ensure that the SIP message contains a Contact: header #If so and client is Nated if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); #Appends parameters to Contact header URI fix_nated_register(); #Adds received IP port to top most via header #This enables subsequent SIP messages to return to the correct port force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- #Is message sender behind NAT? if (client_nat_test("3")) { #If so then setflag7 setflag(7); force_rport(); #Rewrite Contact: header to contain Sip client's public IP:port fix_nated_contact(); }; if (!allow_trusted() && !is_uri_host_local()) { # if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { #Enable mediaproxy if needed route(5); #Send message to destination route(1); return; }; if (uri=~"^sip:(\+|00|011)[0-9]*@") { # International PSTN log(1,"international PSTN match"); route(4); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{11}@") { # Domestic(US) PSTN route(4); return; }; sl_send_reply("404", "User Not Found"); return; }; if (method=="CANCEL") { route(1); return; }; #Set accounting flags setflag(1); setflag(3); route(5); log(1,"route(5) called from from route(3)"); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- #UAC module routing# t_on_failure("1"); resetflag(9); log(1,"resetflag(9)"); uac_replace_from("sip:645566778@gateway.co.nz"); rewritehost("203.184.100.100"); log(1,"host rewritten"); avp_write("i:45", "inv_timeout"); setflag(1); setflag(3); route(5); log(1,"route(5) called from from route(4)"); route(1); } route[5] { # ----------------------------------------------------------------- # NAT traversal # ----------------------------------------------------------------- #check whether message sender(7) or recipient(6) is Nated if (isflagset(6) || isflagset(7)) { #If so then proxy media log(1,"using media proxy in route(5)"); use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; lookup("location"); route(1); } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { log(1,"hit failure_route[1]"); if (t_check_status("401|407")) { log(1,"status is 401 or 407"); if (isflagset(9)) { log(1,"flag9 is set"); t_reply("503","Authentication failed"); return; } if (uac_auth()) { log(1, "called uac_auth"); setflag(9); t_on_failure("1"); append_branch(); #t_relay(); log(1, "calling route(1) from failure_route"); route(1); } } }

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[Users] User Agent with active/passive direction support
by Ladislav Andel 13 Apr '06

13 Apr '06

Hi list, Does anybody know what UA supports active/passive direction attribute except PSTN, Voicemail gateways? (Regarding Connection Oriented Media) Thanks. Ladislav

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[Serusers] Warning Message
by Suresh Arunachalam 13 Apr '06

13 Apr '06

Hi all, I am using presently ser-0.9.6 and have uac within the ser. whenever i run my application i get this warning Warning: run_failure_handlers: no UAC support (0, 0) I am not much clear about this warning . can someone give me an idea about this. Thanks&Regards A.Suresh -- A.Suresh Pandora Networks

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R: [Users] load from db table
by D'Addelfio Davide 13 Apr '06

13 Apr '06

Hi Bogdan, I'm trying to setup my config file , did the same modparam("avpops", "db_scheme", "scheme0:username_col=from;value_col=timestamp;value_type=string;table=acc") modparam("avpops","avp_aliases","timestamp=i:800") if (method=="INVITE") avp_db_load("$from","$timestamp/$scheme0"); I'm not sure that is correct... log gives me these errors Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: submit_query: You have an error in your SQL syntax. Check the manual that corresponds to your MySQL server version for the right syntax to use near 'from='bob'' at line 1 Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: db_query: Error while submitting query Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: ERROR:avpops:load_avps: db_load failed Any thought? Thanks Davide -----Messaggio originale----- Da: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per conto di Bogdan-Andrei Iancu Inviato: martedì 11 aprile 2006 16.12 A: D'Addelfio Davide Cc: users(a)openser.org Oggetto: Re: [Users] load from db table Hi, see: http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_db_load the "db_scheme" example. regards, bogdan D'Addelfio Davide wrote: >Hi Bogdan, > >i setup my openser.cfg to store SIP messages into acc tables of mysql, >using extra accounting to store also the body part of the message. >Now i need that openser read into that db's table, in particular in some >rows of db. >If I use avp_db_load it works only over usr_preference table, instead I >want it looks into acc table. > >How can I do? > >Thanks for help >Davide > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Users] Question regarding OpenSER behind NATs with NATed clients
by Justin Pearce 13 Apr '06

13 Apr '06

Hi, I figured that if the proxy was behind a NAT that it would need port forwarding. But, if the clients are behind different NATs, is there anything special that needs to be in place, aside from port forwarding on their firewalls? If the clients are able to determine their own public IP addresses, is there anything I still need to do to ensure that INVITEs and BYEs get to the right place (like NATHELPER and RTPPROXY or MEDIAPROXY)? I'm trying to figure which direction I need to go with this before I start to do something, only to find that I needed to go the other way with it. Thanks, Justin Pearce Price Video Productions Justin(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, April 12, 2006 4:06 AM To: Justin Pearce Cc: users(a)openser.org Subject: Re: [Users] Question regarding OpenSER behind NATs with NATed clients Hi, if you place the proxy behind a NAT, I think you need to do a port forwarding for the 5060 port - otherwise the proxy will not be reachable from the public internet. regards, bogdan Justin Pearce wrote: > Greetings, > > I know it is probably a really newbie question, but is OpenSER able to > negotiate the following situation: > > Server running OpenSER sits NATed behind a firewall. Clients sit on a > separate network, NATed behind a firewall. > > Client systems send signaling information to the server to contact > another client. Clients are capable of determining > > their own outside IP addresses. Client then connects media streams > directly to other client. > > I'm trying to find out of this is possible and, if so, where might > documentation for it be found, since my supervisor is > > asking me about it and I can't seem to find any relevant information > on the topic (maybe my googling skills are not up > > to par). > > Any info would be greatly appreciated. > > Thanks, > > Justin Pearce > > Price Video Productions > > Justin(a)PriceVideo.com <mailto:Justin@PriceVideo.com> > > 361-572-3810 > > 800-733-3810 > > Fax: 361-572-3894 > > www.PriceVideo.com > >----------------------------------------------------------------------- - > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Serusers] User Agent with active/passive direction support
by Ladislav Andel 13 Apr '06

13 Apr '06

Hi list, Does anybody know what UA supports active/passive direction attribute except PSTN, Voicemail gateways? Thanks. Ladislav

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Re: [Users] registrar table size
by Bogdan-Andrei Iancu 13 Apr '06

13 Apr '06

Hi Kent, first of all, you cannot have partial caching into memory - usrloc either cache all db content ,either nothing (that's the design). now, by size of the register table I understood the size of the hash table used by usrloc - number of entries the hash has; but this does not affect how many records can be hold into usrloc table. so, what you are trying to do is not possible. regards, bogdan Poppenhouse Kent-P28924 wrote: >Thank you for your quick reply. I have been out of the office all last >week so I was just able to try it out. I changed the new_udomain to 2 >in the dlist.c file. It didn't seem to change anything. I was still >about to register 8000 users. Let me explain what I'm trying to do, >maybe you can steer me the correct way. > >I have a mysql database created which the SER successfully updates with >the user registrations. What I would like to have happen is have >100,000 users registered with the SER storing only 1000 users in memory >and performing database lookups to find the location of the other users. >These numbers are just made up but is something like this possible? I >tried to register users yesterday and when it had registered 78682 it >said it ran out of memory. How can I get the SER to store fewer users >in memory but still update the database for future lookups? > >Thank you for any help you can provide. > >Kent > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Friday, March 31, 2006 1:35 AM >To: Poppenhouse Kent-P28924 >Cc: users(a)openser.org >Subject: Re: [Users] registrar table size > >Hi, > >the size of the hash table is hardcoded (no config option :( ). See >modules/usrloc/dlist.c line 341, the call of new_domain() - second param > >is the size. >Note that the size must be a power of 2! > >regards, >bogdan > >Poppenhouse Kent-P28924 wrote: > > > >>Does anyone know how to change the size of the location table used for >> >> > > > >>the registrar? I would like to reduce it from the default 512. >> >>----------------------------------------------------------------------- >> >> >- > > >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > >

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[Users] How to install and work on SerWeb in openSER
by raviprakash sunkara 13 Apr '06

13 Apr '06

Hello, Currnently now am working on SER by openser... I installed oenser-1.0.1.rhel3.i386 into my box. openser server is workiing fine. I want how to install and config the serweb in openSER please help it....... Thanks & Regards -- Sunkara Ravi Prakash.. hyperion Technologies.....

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AW: [Users] Allow only TLS connections
by Thorsten.Haupt＠t-systems.com 13 Apr '06

13 Apr '06

I searched for this function, but I didn't found it :-( Knows anyone the correct code, not only pseudo-code? Torsten -----Ursprüngliche Nachricht----- Von: Cesc [mailto:cesc.santa@gmail.com] Gesendet: Dienstag, 11. April 2006 14:03 An: Haupt, Thorsten Cc: users(a)openser.org Betreff: Re: [Users] Allow only TLS connections I think in openser there is a function to check what transport the message came in ... you can do something like: if ( transport != TLS ) { send error to UA break; } Cesc On 4/11/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > > > Hello, > > I use OpenSER in a testing environment for VoIP security. My clients > connect via TLS. If I deactivate UDP/5060 on the server, it doesn't work correct. > Some Clients can't connect and others can't establish calls. I read in > another thread, that UDP is mandatory for SIP and that the server need it. > > But how can I prevent users from connecting via UDP and force them to > use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But is > this the correct way? Are there any parameters server-side to force > users to connect via TLS? > > Thanks for response. > Torsten > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] "> /dev/stderr" versus ">&2"
by Nick Hoffman 13 Apr '06

13 Apr '06

Hi guys, I noticed that in /usr/sbin/serctl , output is redirected to stderr this way: > /dev/stderr rather than this way: >&2 I'm curious as to why "> /dev/stderr" was chosen rather than the usual ">&2". Any thoughts? Cheers, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Serusers] redundant ser
by Thomas Britis 13 Apr '06

13 Apr '06

Hi all, I have two machines running ser. One is just a backup and assumes operation when (and until) the main server is down. Now I'd like to have one more server, but not as backup but as a round robin server. What's the best way to do that ? I was thinking of replicate the mysql db but I didn't know what could happen to location the location table (one registered on server1 should be able to invite a user on server2 and vice-versa). Thank you in advance. -- Thomas Storino Britis TCNet Informatica e Telecomunicacoes LTDA

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[Serusers] Problem with Voice Quality
by mkumar＠mantragroup.com 13 Apr '06

13 Apr '06

Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges between 10 to 60 ms delay but the average is near to 20 ms. We use SIP. How can we solve this problem, is there any setting at the server end to handle this, as clients have very limited resources we have to manage this at the server end, please tell me how can I do this? Thanks, Manoj.

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[Serusers] Shoot my idea full of holes, please.
by sip 12 Apr '06

12 Apr '06

Okay... I've been hacking about with a way to handle pre-paid dialout stuff with Asterisk-b2bua and sheer, ugly hacking. I'll explain. Asterisk-b2bua basically is code that sets the session timer of an asterisk dial session based on a value it pulls out of the radius DB. Now... the trick is, of course, that something has to SET that value for the call, and then you're accessing the radius server without a password to do authentication. Passwordless radius(only authentication -- no authorisation), grabbing its data from a DB, is a bit like grabbing data directly from the DB... but... well... with something in the way. So... something has to set the session timer based on the price of the call divided into the amount of money a person might have in his account. As opposed to writing a script to JUST set a session timer (and creating an additional DB store/lookup), I've hacked the b2bua.pl script to skip the radius authentication and hit the DB direct for authentication in a billing DB. From that billing DB, it then grabs the amount of money left in a person's account... then goes through the rate file (which I've converted to a sorted jagged array for optimisation) to determine which prefix is a best match for that call, and grabs the amount per minute that the call costs. It does a quick divide to determine how many minutes that would allow for a call, converts it to seconds, and then sets the session timer based on the seconds available for a call, using the surrounding code of the b2bua.pl script. At the end of the script, it calculates the time on the call either before the timer expired or after and updates the amount in that person's account accordingly. SO... that's the logic behind the overall hack. Since I'm barely awake and have been staring at this code for something like 100+ hours now (including time editing the rate file for optimisation), I can barely see what I'm typing, much less see any gaping logical errors I may have made. Does anyone see any HUGE, insurmountable problems with my idea? Any small, perhaps surmountable problems? Any possible issues I need to watch out for (other than not allowing more than one outgoing PSTN call per account at a time)? If you do, I'd love to hear them. Thanks. N.

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[Serusers] sl_send_reply: I won't send a reply for ACK!!
by sip 12 Apr '06

12 Apr '06

I've gone through my ser.cfg and actually REMOVED every single sl_send_reply in an attempt to figure out where I've got these stray ACK messages being sent a reply, but even with NO sl_send_reply commands in there, I get that cursed message in the logs -- usually when dealing with reinvites (which I appear to also be having some issues with atm). Is there something ELSE... some module or some such that might be causing this error other than a straight sl_send_reply command? N.

1 0

[Serusers] 477 Message by SER???
by Pat wang 12 Apr '06

12 Apr '06

I am using SER 0.9.6 and just saw a strange 477 message sent by SER. When an INVITE arrives to SER, the SER is not able to find default route to proxy the INVITE according to the RURI and SER reply a 477. The SIP RFC does not define such a message. Is this SER's own "protocol"? This is the RURI in the INVITE to SER: INVITE sip:3205@192.170.12.11 SIP/2.0 While there is no IP route to 192.170.12.11 on the Linux that running SER, SER reply: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Ya, the problem can be easily fixed by adding a route to that 192.170.12.x subnet (for peering). But I realy want to know how does this 477 come from? Anyone? Thanks, Pat

2 1

RE: [Serusers] cisco sip phones?
by Gould, Aaron 12 Apr '06

12 Apr '06

copy and paste. darnit thanks jan.......now i know who to come to for cisco voice related questions ;) aaron ________________________________ From: Jan Bollen [mailto:jbollen@cisco.com] Sent: Wed 4/12/2006 6:27 AM To: Gould, Aaron Cc: Cesc; SER-Users Subject: RE: [Serusers] cisco sip phones? On Wed, 12 Apr 2006 Gould, Aaron kneaded and moulded: | |oh, cisco phone, ummm, yeah i put the 5 files in a tftp server i think it was ... | |OS79XX |P003-08-2-00.bin |P003-08-2-00.sbn |P003-08-2-00.loads |P003-08-2-00.sb2 which is definitely not correct: - if you're running version 7 or higher there is no need for the OS79XX.TXT file - the .bin and .sbn are the unsigned and signed images containing the universal application loader. - the .loads and .sb2 files must be coded P0S3... and not P003. I know, it is confusing, but that is the way it is. cheers, Jan | |and it loaded up a new universal application loader into the phones....."...2000-2006" | |then they loaded their sip firmware ....blinky lights during upgrade.....now they boot up and go through a decision tree to find a valid config file which i haven't been able to figure out yet...... | |aaron | |________________________________ | |From: serusers-bounces(a)lists.iptel.org on behalf of Cesc |Sent: Wed 4/12/2006 6:07 AM |To: SER-Users |Subject: [Serusers] cisco sip phones? | | | |Hi, | |I am looking for experiences with cisco (series 79xx) phones with sip |... can it be installed (sip) on all of them? or just a few models |(7960 ... or 7970)? | |Tks! | |Cesc | |_______________________________________________ |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 |

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[Users] CRL in OpenSER
by Christoph Fürstaller 12 Apr '06

12 Apr '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Is there a possibility to check a client certificate against a CRL? Is this allready implemented or are there planes to do such? Is it a good idea to use client certs? Or is the effort to realice that to much? Cause the benefits from authenticating a client only for the TLS connection isn't that much. And authentication against a DB is done later on in OpenSER as well. (authentication is done twice) What do you think? chris... -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEPPHbR0exH8dhr/YRAncQAJ9IEd6eO4cxgeoIna39VwAKnCoz9QCeNEtr AjCFWx/cTjDcUBBe+EvBQFs= =fZHN -----END PGP SIGNATURE-----

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RE: [Serusers] loading solaris 10 and ser again
by Gould, Aaron 12 Apr '06

12 Apr '06

do you have the exact link? i'm there looking at it but see a lot. is it one of these listed below or is it somewhere else? thanks Top Downloads All Config FIles f... (2162) Complete source pa... (1964) SER 0.9.7-pre1 sou... (1631) Issue 05 - Getting... (1334) Issue 5.0 - Gettin... (994) SER GettingStarted... (923) SER Getting Starte... (551) Hennreich & Johnst... (539) LCR module (442) xlog module (186) Top Links AVPops Module documentat... (976) sip-router.org (821) UAC module Documentation (504) Voice-system SER CVS Sna... (471) SIP.edu : Voice over IP ... (411) IPClouds (311) tech-invite (280) Free IP Call (257) TeleAppliant ITSP (207) Iptelorg.com (185) Recent Downloads LCR working with m... (2006/4/4) SER 0.9.7-pre1 sou... (2006/2/13) Complete source pa... (2006/2/13) SER GettingStarted... (2006/2/13) OSP Peering Module... (2006/2/13) Issue 5.0 - Gettin... (2005/9/12) All Config FIles f... (2005/9/12) Issue 05 - Getting... (2005/9/12) SER Getting Starte... (2005/5/3) Hennreich & Johnst... (2005/4/5) Recent Links Freespeech.ie - Irelands... (2006/3/3) Operations & Billing Sup... (2006/1/24) ConexionGroup SIP Trunk ... (2005/10/2) tech-invite (2005/6/24) Free IP Call (2005/6/15) IPClouds (2005/5/19) UAC module Documentation (2005/4/19) Voice-system SER CVS Sna... (2005/4/12) Telecommunications Consu... (2005/4/12) TeleAppliant ITSP (2005/4/5) ________________________________ From: Simon [mailto:sermail@systemsrm.co.uk] Sent: Wed 4/12/2006 6:04 AM To: Gould, Aaron; serusers(a)lists.iptel.org Subject: RE: [Serusers] loading solaris 10 and ser again Aaron, Not sure about advice for the Solaris 10, but the onsip.org site has a Getting Started guide that will provide a basic ser setup for you as well as some more complex configurations. Simon -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Gould, Aaron Sent: 12 April 2006 14:01 To: serusers(a)lists.iptel.org Subject: [Serusers] loading solaris 10 and ser again i'm re-running the install for solaris 10 again. when i rerun the setup instructions for setting up ser is there any advice you more seasoned ser folks could share with me? http://developers.sun.com/solaris/articles/solaris_as_sip/solaris_as_sip.htm l i would appreciate any tips you would provide for things i might run into. also, does ser function as a sip registrar, proxy and redirect server by default? if so, i'm wondering why when my softphones sent register messages to it they were simply forwarded back to the sip phone. ? aaron _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] cisco sip phones?
by Cesc 12 Apr '06

12 Apr '06

Hi, I am looking for experiences with cisco (series 79xx) phones with sip ... can it be installed (sip) on all of them? or just a few models (7960 ... or 7970)? Tks! Cesc

2 1

[Serusers] loading solaris 10 and ser again
by Gould, Aaron 12 Apr '06

12 Apr '06

i'm re-running the install for solaris 10 again. when i rerun the setup instructions for setting up ser is there any advice you more seasoned ser folks could share with me? http://developers.sun.com/solaris/articles/solaris_as_sip/solaris_as_sip.ht… i would appreciate any tips you would provide for things i might run into. also, does ser function as a sip registrar, proxy and redirect server by default? if so, i'm wondering why when my softphones sent register messages to it they were simply forwarded back to the sip phone. ? aaron

2 1

[Users] Gateway authentication
by Daniel Korndorfer 12 Apr '06

12 Apr '06

Hi! Has anyone tried to use gateway authentication with the lcr module? Tks, D.K.

2 3

[Serusers] serweb tunning :)
by Joao Pereira 12 Apr '06

12 Apr '06

Hello to all in the subscription page of serweb ( serweb/user/reg/index.php ) the users can choose their timezone from a list. But 95% of my users have the "Europe/Lisbon" timezone. How can I put as default timezone "Europe/Lisbon", instead of the " - select your timezone -" ? Thanks Joao Pereira

3 3

[Serusers] snmp module
by hgaillac-sip＠yahoo.fr 12 Apr '06

12 Apr '06

hello, Is there a project to monitor ser via snmp ? Is http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/obsolete/snmp/ died ? Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

2 2

Re: [Serusers] chown'ing /tmp/ser_fifo
by Kim Culhan 12 Apr '06

12 Apr '06

On Wed, April 12, 2006 1:25 am, Nick Hoffman said: > Hi guys, I need /tmp/ser_fifo to *always* be owned by root:foobar. What's > the best way of accomplishing this? You can use this in ser.cfg: fifo_group=www fifo_mode=0660 # fifo's permissions -kim -- w8hdkim(a)gmail.com

1 0

[Users] one way audio problem
by unplug 12 Apr '06

12 Apr '06

I have an openser using mediaproxy with port setting 2000:8000. Most of the user can use it without problem. But few user experiences one way audio when make a call to the called party. Is it related to the setting of mediaproxy (mediaproxy.ini)? Is it related to the port range I set below? portRange = 2000:8000 Below is the information that I capture from the location table. One is using a hardware IP phone but having one way audio problem. One is using XLite without problem. Both client are in the same network behind NAT to connect to the same openser server. case 1: using hardware IP phone with one way audio in making a call. The user claims he is using NAT with internal IP 192.xxx.xxx.xxx. But there is no such information shown in the locatioin table. location table information ----------------------------- contact = sip:1234@202.123.12.1:5060 received = sip:202.123.12.1:60626 callid = TtFZUeEbHpx4suRG(a)202.123.12.1 cseq = 3834 case 2: using XLite software phone without problem. location table information ----------------------------- contact = sip:sip:2345@202.123.12.1:47328 received = sip:202.123.12.1:47328 callid = 56725116a22ec84f@UmF5bW9uZC1wYy5waC5vd3RlbC5jb20. cseq = 11 I wonder why there is one way audio problem only for the case 1 but not for case 2. I do think there is setting problem but I can't figure out. Anyone can help? system: openser 1.0.1 + mediaproxy 1.4.2

1 0

[Users] Overlapping AVPs
by Andreas Granig 12 Apr '06

12 Apr '06

Hi, Me again, sorry, but the docs aren't really noisy about AVP details... So if I have user preferences for both the caller and callee and load them from DB and print them using the following: avp_db_load("$avp($uuid_caller)", ""); avp_db_load("$avp($uuid_callee)", ""); avp_print(); then they may overlap because of the same ID (say "i:102" for toggling some specific feature on/off), but according to the debug output both are present: INFO:avpops:print_avp: p=0x4056db90, flags=100 INFO: id=<102> INFO: val_int=<1> INFO:avpops:print_avp: p=0x4056dc68, flags=100 INFO: id=<102> INFO: val_int=<0> So is it possible to selectively access the avp-value of both $uuid_caller and $uuid_callee? Something like $avp(i:102)[0] and $avp(i:102)[1] maybe? Thanks, Andy

2 5

RE: [Fwd: Re: [Users] fax support]
by unplug 12 Apr '06

12 Apr '06

Does openser support T.38? Does it enable by default? >The problem with G.711 + FAX is concerned with variable delay and jitter... >Use T.38 or V.150 if enable... ;) >Edson.

2 1

[Users] Question regarding OpenSER behind NATs with NATed clients
by Justin Pearce 12 Apr '06

12 Apr '06

Greetings, I know it is probably a really newbie question, but is OpenSER able to negotiate the following situation: Server running OpenSER sits NATed behind a firewall. Clients sit on a separate network, NATed behind a firewall. Client systems send signaling information to the server to contact another client. Clients are capable of determining their own outside IP addresses. Client then connects media streams directly to other client. I'm trying to find out of this is possible and, if so, where might documentation for it be found, since my supervisor is asking me about it and I can't seem to find any relevant information on the topic (maybe my googling skills are not up to par). Any info would be greatly appreciated. Thanks, Justin Pearce Price Video Productions Justin(a)PriceVideo.com <mailto:Justin@PriceVideo.com> 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com

2 1

[Users] Force RTP stream to go through mediaproxy
by Sam Lee 12 Apr '06

12 Apr '06

Hi all, I got the openser and mediaproxy up working and fine. The mediaproxy is able to work with certain NATED configurations, but not all. When i tried to check what is the problem , those NATED configurations that were not working was found not to have made use of the mediaproxy, which causes the problem. What i am trying to do now, but without much success, is to force all the RTP media to go through the mediaproxy. Is there any way i can force all the RTP media to go through the mediaproxy without all the client_nat_test stuff...? Any hints will be much appreciated ! Regards, Sam

2 1

[Users] small typo on the mangler doc
by Ovidiu Sas 12 Apr '06

12 Apr '06

The doc for mangler module shall be updated: -modparam("module", "contact_flds_separator", "-") +modparam("mangler", "contact_flds_separator", "-") see http://openser.org/docs/modules/1.0.x/mangler.html#AEN55 -ovi

2 1

[Users] Media Proxy Problem
by Rafael Neves Harff 12 Apr '06

12 Apr '06

Hi all! Ive finished installing Openser + mediaproxy today. I have user Ozan Blotter post in the list (http://openser.org/pipermail/users/2005-July/000304.html) to configure the mediaproxy.It worked before without media proxy. Now I can authenticate, the same way I did before, but when I try to dial someone I get a message in the Softphone: Call failes: 407 Proxy Authentication Required What I'm doing wrong? Thanks for the help! --- Rafael Neves Harff from Brazil

2 1

[Users] function client_nat_test problem
by unplug 12 Apr '06

12 Apr '06

below is the configuration Internet --- openser (mediaproxy) ---- PSTN + public IP (203.13.14.210) + Internal IP (10.0.0.30) openser will list on both internal and external IP User can make/receive call without problem. user device --- NAT -------------------------------------- openser ---- PSTN (10.0.0.60) (202.130.10.200) (203.13.14.210) user device ------------------------------------------------ openser ---- PSTN (10.0.0.60) (10.0.0.30) User can receive call without problem. However, when user makes outgoing calls, problem will occur. 1. called party will ring but calling party without ring. 2. After pick up the phone, both parties can't hear any sound. >From the past experience, the problem should be caused by the NAT issue. In my dial plan, I use a module function client_nat_test("3"). if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.13.14.210:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; >From the function description http://openser.org/docs/modules/devel/mediaproxy.html#AEN113, it will test if the client is behind NAT or not. However, in the result of ngrep, I can still find the header with the following. Record-Route: <sip:203.13.14.210:5060;nat=yes;ftag=nQvg2qq2fzNg24d1;lr=on> That's mean function client_nat_test always return true (from the case above) no matter client is behind NAT. It is what I find up to now. I would like to ask whether there is any other configuration variables needed for the NAT detection. How the mediaproxy module (based on what) to detect the device is behind NAT or not? Any clue is welcome.

2 1

[Users] Upgrade to 1.0.1 and usrloc alias isn't working.l
by Shane Burrell 12 Apr '06

12 Apr '06

I just upgraded to v 1.0.1 from 0.9.5 and now calls can be made from sip devices but calls coming from PSTN don't find a usrloc or alias and its sent back out by 10 digit check. I'm using MYSQL. What and were should I look for setting to make the following call work to a particular sip user from the TNT. lookup(): '8485974202(a)102.53.11.19' Not found in usrloc The TNT is sending the call to '8485974202(a)102.53.11.19' which appears to be correct in the alias file for user shane who has an entry in the location table. TIA, Shane ***************************************************************

2 1

[Users] multiproxy routing question
by HUANG Quan 12 Apr '06

12 Apr '06

Hi all, I wanna setup a simple sip voip system with two proxies, namely proxy1 and proxy2. Proxy1 handles the phone numbers between 1001 and 1999, and proxy2 handles the numbers between 2001 and 2999. Could anybody give me an example script to route sip messages between 1xxx and 2xxx?

1 0

Re: [Users] Fwd:Invalid table version
by buxalex＠libero.it 12 Apr '06

12 Apr '06

Hello Bogdan, as usual you are right! :o) I tried to use the table subscriber as follow: if (method=="INVITE") { setflag(2); record_route(); if (!lookup("subscriber")) { sl_send_reply("401", "Dialled Number not Authorizied"); exit; if (!lookup("location")) { sl_send_reply("404", "Dialled number off line !"); exit; }; }; Of course commenting the part related to subscriber table it works fine. My idea was to check whether the dialled number is only off line or it is not existent; this because I would like to have the choice to send a registered vocal message (or for mow an email) to dialled numbers. I tried to put 1003 in table domain and subscriber, but now I get this error: ACC - initializing AVPops - initializing submit_query: Unknown column 'contact' in 'field list' db_query: Error while submitting query preload_udomain(): Error while doing db_query register_udomain(): Error while preloading domain 'subscriber' domain_fixup(): Error while registering domain fix_actions: fixing failed (code=-1) at cfg line 211 ERROR: fix_expr : fix_actions error Can I fix this problem with configuration, or should I modify the logic to access the db (I mean I shouldn't use the subscriber table)? Thanks a lot Alessandro ---------- Initial Header ----------- >From : "Bogdan-Andrei Iancu" bogdan(a)voice-system.ro To : "buxalex(a)libero.it" buxalex(a)libero.it Cc : "users" users(a)openser.org Date : Tue, 11 Apr 2006 20:34:58 +0300 Subject : Re: [Users] Fwd:Invalid table version > Hi, > > the error is generated by the usrloc module - the used tables must have > version 1003; are you using a different tablet then "location" or > "aliases"? if not, check in version table their records. > > regards, > bogdan > > buxalex(a)libero.it wrote: > > >Hello, > >I tried also to drop and the to install the db, but still I get the same error.... > > > >Please anybody can help me? > > > >I am using openser version 1.1.0 dev17 tls > > > >thanks in advance > >Alessandro > >---------------------------------------------------------------- > >Hello, > > > >running openser with mysql db (through monitor), I retrieve: > > > >ACC - initializing > >Apr 11 18:58:32 dhc27-53-93pc /usr/local/sbin/openser[7806]: register_udomain(): Invalid table version (use openser_mysql.sh reinstall) > > > >If I comment the module ACC loading I get the same error for the other modules. > >I reinstalled, as suggested from the log the db, but still I get the error. > > > >in table version I can see that the domain table has version 1 > > > >Any idea? This is a test db, do you suggest to erase it and to install it again? > > > >thanks a lot > >Alessandro > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

2 1

[Serusers] serweb configuration
by Garima Mishra 12 Apr '06

12 Apr '06

Hi all, I am trying to install serweb but through web browser i am getting following fatal error, Fatal error: Failed opening required 'XML/RPC.php' (include_path='.:/usr/share/pear') Can someone please help me to rectify this error? Thanks, Garima Mishra Software Engineer <http://www.openeratech.com/> <http://www.openeratech.com/> <http://www.openeratech.com/> Openera Technologies Pvt Ltd. 97, Block 5 Koramangala Indl. Estate <?xml:namespace prefix = st1 ns = "urn:schemas-microsoft-com:office:smarttags" />Bangalore - 560 095. India Tel: +91 80 2552 1937 Fax: +91 80 2552 1164 <http://www.openeratech.com/> www.openeratech.com E-mail: <mailto:garima@openeratech.com> garima(a)openeratech.com This message (including any attachments) is a confidential and Privileged communication to the intended addressee. If you are not the Intended addressee, you have received this message in error. In that case, please permanently delete this message and notify the sender so that we can avoid this inconvenience in the future. Thank you.

3 2

Re: [Users] nat_helper: multiple media IP address in SDP
by Dmitry Lyubimkov 12 Apr '06

12 Apr '06

We had the same problem with SDP. There are very many UA with this mistake error To bypass this restriction we have added after force_rtp_proxy(); Also such command subst("/^c=IN IP4 ([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/c=IN IP4 11.22.33.44\5/i"); Dmitry ------------------------------ Message: 6 Date: Tue, 11 Apr 2006 16:51:22 +0200 From: "Nicolas Olivier" <nolivier(a)alphalink.fr> Subject: Re: [Users] nat_helper: multiple media IP address in SDP To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> Cc: users(a)openser.org Message-ID: <443BC26A.8040407(a)alphalink.fr> Content-Type: text/plain; format=flowed; charset="ISO-8859-1" Ok, I may have a look to the csv. Thanks for the help. regards, Nicolas Bogdan-Andrei Iancu wrote: > Hi, > > Nicolas Olivier wrote: > > > > > Hi Bogdan, > > > > Ok, I understand now. But I still encounter the problem because: > > - rtpproxy only rewrites the c= from media part (but it should be fine > > as you said) despite what a quick look in the rtpproxy code comments > > say ("We have to change ports in m-lines, and also change IP addresses > > in c-lines which can be placed either in session header (fallback for > > all medias) or media description.") > > yes, the nathelper will change the c= from session header only if it > finds a media section without a local c= (which means the default c= > from session hdr will be used). > > > - the centrex (which is an asterisk by the way) take only into account > > the c= from the session part, not the one from media part > > in the CVS devel there is a flag that force also changing of session c= : > http://openser.org/docs/modules/1.1.x/nathelper.html#AEN275 , the > "c" flag > > regards, > bogdan > > > > > > > > > regards, > > Nicolas > > > > Bogdan-Andrei Iancu wrote: > > > >> Hi Nicolas, > >> > >> it;s perfectly ok - see the SDP RFC : an SDP may contain a default > c= in > >> the session part; each media section (m=) may contain an ip (c=); if it > >> doesn't the session c= will be used. > >> > >> regards, > >> bogdan > >> > >> Nicolas Olivier wrote: > >> > >> > > >> > Hi, > >> > > >> > I've got a gateway which is only used for rounting and rtp proxying > >> > between providers and centrexes. > >> > > >> > On reply to an INVITE, one of our provider send back a "183 Session > >> > Progress". The problem is that in the SDP block, we've got 2 > media IP > >> > address and rtpproxy only rewrite one. > >> > > >> > Finally, the provider establish rtp session with our gateway, and > our > >> > centrex directly with the provider. > >> > > >> > provider gateway centrex > >> > 172.16.0.10 192.168.1.10 192.168.1.20 > >> > RTP -------------> RTP ------------> RTP > >> > ^-------------------------------------------------| > >> > > >> > So my questions are, is it possible to have multiple IP address in > >> SDP > >> > and if so, how can I tell rtpproxy to rewrite all of them. > >> > > >> > Coming from provider: > >> > > >> > SIP/2.0 183 Session Progress. > >> > Via: SIP/2.0/UDP > >> > 192.168.1.10;branch=z9hG4bKdd67.a4cc2c44.0,SIP/2.0/UDP > >> > 192.168.1.20:5062;branch=z9hG4bKdd67.08f45a33.0,SIP/2.0/UDP > >> > 192.168.1.20:5060;branch=z9hG4bK4af242b7. > >> > From: "02" <sip:0143132445@192.168.1.20>;tag=as226ce7b9. > >> > To: <sip:0123456789@192.168.1.20:5062>;tag=3123AAA8-20C5. > >> > Date: Tue, 11 Apr 2006 09:10:29 GMT. > >> > Call-ID: 079ab6663e403ff44a1107e5111b075f(a)192.168.1.20. > >> > Server: Cisco-SIPGateway/IOS-12.x. > >> > CSeq: 102 INVITE. > >> > Allow-Events: telephone-event. > >> > Contact: <sip:677238#0123456789@172.16.0.10:5060>. > >> > Record-Route: > >> > > >> > <sip:192.168.1.10;ftag=as226ce7b9;lr=on>,<sip:192.168.1.20:5062;ftag=as2 26ce7b9;lr=on>. > > >> > >> > > >> > Content-Disposition: session;handling=required. > >> > Content-Type: application/sdp. > >> > Content-Length: 261. > >> > . > >> > v=0. > >> > o=CiscoSystemsSIP-GW-UserAgent 3448 4768 IN IP4 172.16.0.10. > >> > s=SIP Call. > >> > c=IN IP4 172.16.0.10. > >> > t=0 0. > >> > m=audio 18322 RTP/AVP 18 101. > >> > c=IN IP4 172.16.0.10. > >> > a=rtpmap:18 G729/8000. > >> > a=fmtp:18 annexb=no. > >> > a=rtpmap:101 telephone-event/8000. > >> > a=fmtp:101 0-16. > >> > > >> > Forwarded to centrex: > >> > > >> > SIP/2.0 183 Session Progress. > >> > Via: SIP/2.0/UDP > >> > 192.168.1.20:5062;branch=z9hG4bK43a4.3e96aba3.0,SIP/2.0/UDP > >> > 192.168.1.20:5060;branch=z9hG4bK3213db83. > >> > From: "02" <sip:0143132445@192.168.1.20>;tag=as1a2f900d. > >> > To: <sip:0123456789@192.168.1.20:5062>;tag=3121D1B4-1BFD. > >> > Date: Tue, 11 Apr 2006 09:08:28 GMT. > >> > Call-ID: 08467c5e299ab833106517c63d3edc2e(a)192.168.1.20. > >> > Server: Cisco-SIPGateway/IOS-12.x. > >> > CSeq: 102 INVITE. > >> > Allow-Events: telephone-event. > >> > Contact: <sip:677238#0123456789@172.16.0.10:5060>. > >> > Record-Route: > >> > > >> > <sip:192.168.1.10;ftag=as1a2f900d;lr=on>,<sip:192.168.1.20:5062;ftag=as1 a2f900d;lr=on>. > > >> > >> > > >> > Content-Disposition: session;handling=required. > >> > Content-Type: application/sdp. > >> > Content-Length: 277. > >> > . > >> > v=0. > >> > o=CiscoSystemsSIP-GW-UserAgent 565 174 IN IP4 172.16.0.10. > >> > s=SIP Call. > >> > c=IN IP4 172.16.0.10. > >> > t=0 0. > >> > m=audio 36296 RTP/AVP 18 101. > >> > c=IN IP4 192.168.1.10. > >> > a=rtpmap:18 G729/8000. > >> > a=fmtp:18 annexb=no. > >> > a=rtpmap:101 telephone-event/8000. > >> > a=fmtp:101 0-16. > >> > a=nortpproxy:yes. > >> > > >> > > >> > openser.cfg > >> > > >> > (...) > >> > > >> > onreply_route[1] { > >> > if (status =~ "(180)|(183)|2[0-9][0-9]") { > >> > fix_nated_contact(); > >> > if (!search("^Content-Length:[ ]*0")) { > >> > force_rtp_proxy(); > >> > }; > >> > } else if (nat_uac_test("1")) { > >> > fix_nated_contact(); > >> > }; > >> > } > >> > > >> > (...) > >> > > >> > Best regards, > >> > Nicolas Olivier > >> > > >> > > >> > _______________________________________________ > >> > Users mailing list > >> > Users(a)openser.org > >> > http://openser.org/cgi-bin/mailman/listinfo/users > >> > > >> > > > ------------------------------ Message: 7 Date: Tue, 11 Apr 2006 16:52:14 +0200 From: Cesc <cesc.santa(a)gmail.com> Subject: Re: [Users] Allow only TLS connections To: "Thorsten.Haupt(a)t-systems.com" <Thorsten.Haupt(a)t-systems.com> Cc: users(a)openser.org Message-ID: <ce8208420604110752i733143d8k5b565ac45b0cfda2(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook&DokuWiki=6 c17b007ea61fa37b86b391ce1b2a80f#tcp On 4/11/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > I searched for this function, but I didn't found it :-( > Knows anyone the correct code, not only pseudo-code? > > Torsten > > -----Ursprьngliche Nachricht----- > Von: Cesc [mailto:cesc.santa@gmail.com] > Gesendet: Dienstag, 11. April 2006 14:03 > An: Haupt, Thorsten > Cc: users(a)openser.org > Betreff: Re: [Users] Allow only TLS connections > > I think in openser there is a function to check what transport the message came in ... you can do something like: > if ( transport != TLS ) { > send error to UA > break; > } > > Cesc > > On 4/11/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > > > > > > Hello, > > > > I use OpenSER in a testing environment for VoIP security. My clients > > connect via TLS. If I deactivate UDP/5060 on the server, it doesn't work correct. > > Some Clients can't connect and others can't establish calls. I read in > > another thread, that UDP is mandatory for SIP and that the server need it. > > > > But how can I prevent users from connecting via UDP and force them to > > use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But is > > this the correct way? Are there any parameters server-side to force > > users to connect via TLS? > > > > Thanks for response. > > Torsten > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > ------------------------------ Message: 8 Date: Tue, 11 Apr 2006 17:16:49 +0200 From: Andreas Granig <andreas.granig(a)inode.info> Subject: [Users] Overlapping AVPs To: users(a)openser.org Message-ID: <443BC861.6040303(a)inode.info> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, Me again, sorry, but the docs aren't really noisy about AVP details... So if I have user preferences for both the caller and callee and load them from DB and print them using the following: avp_db_load("$avp($uuid_caller)", ""); avp_db_load("$avp($uuid_callee)", ""); avp_print(); then they may overlap because of the same ID (say "i:102" for toggling some specific feature on/off), but according to the debug output both are present: INFO:avpops:print_avp: p=0x4056db90, flags=100 INFO: id=<102> INFO: val_int=<1> INFO:avpops:print_avp: p=0x4056dc68, flags=100 INFO: id=<102> INFO: val_int=<0> So is it possible to selectively access the avp-value of both $uuid_caller and $uuid_callee? Something like $avp(i:102)[0] and $avp(i:102)[1] maybe? Thanks, Andy ------------------------------ Message: 9 Date: Tue, 11 Apr 2006 17:54:56 +0200 From: "D'Addelfio Davide" <Davide.D'Addelfio(a)italtel.it> Subject: R: [Users] load from db table To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> Cc: users(a)openser.org Message-ID: <82C94EFCF026F74EB91A2048B1C963A504065277(a)BESONE.corp.dom> Content-Type: text/plain; charset="iso-8859-1" Hi Bogdan, I'm trying to setup my config file , did the same modparam("avpops", "db_scheme", "scheme0:username_col=from;value_col=timestamp;value_type=string;table=a cc") modparam("avpops","avp_aliases","timestamp=i:800") if (method=="INVITE") avp_db_load("$from","$timestamp/$scheme0"); I'm not sure that is correct... log gives me these errors Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: submit_query: You have an error in your SQL syntax. Check the manual that corresponds to your MySQL server version for the right syntax to use near 'from='bob'' at line 1 Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: db_query: Error while submitting query Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: ERROR:avpops:load_avps: db_load failed Any thought? Thanks Davide -----Messaggio originale----- Da: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per conto di Bogdan-Andrei Iancu Inviato: martedм 11 aprile 2006 16.12 A: D'Addelfio Davide Cc: users(a)openser.org Oggetto: Re: [Users] load from db table Hi, see: http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_db_load the "db_scheme" example. regards, bogdan D'Addelfio Davide wrote: >Hi Bogdan, > >i setup my openser.cfg to store SIP messages into acc tables of mysql, >using extra accounting to store also the body part of the message. >Now i need that openser read into that db's table, in particular in some >rows of db. >If I use avp_db_load it works only over usr_preference table, instead I >want it looks into acc table. > >How can I do? > >Thanks for help >Davide > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users ------------------------------ _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 11, Issue 27 *************************************

2 1

[Serusers] About the Inter SIP Proxy Problem
by AR. 12 Apr '06

12 Apr '06

Dear ALL I have two SIP Proxy servers , One is SER, another one is CSPS (cisco) When I call a phone number that is belong to the CSPS routing, using the SER subscriber, the packet form SER to CSPS will get a problem with "400 Bad request" . (It's ok, If the UA in CSPS call to the UA in SER) My architecture is : UA1----SER----B2BUA---RADIUS | | Internet | | CSPS--UA2 Logs of B2BUA : 20060412-144443.276 [57396144] WARN bcWorker: uaState/UaStateRinging.cxx:133 UaStateRinging::Received 400 20060412-144443.276 [57396144] INFO bcWorker: controlState/StateCallReady.hxx:98 StateCallReady::bye was called, incomplete transaction Does anyone know how to solve it? thanks AR.

2 1

[Serusers] chown'ing /tmp/ser_fifo
by Nick Hoffman 12 Apr '06

12 Apr '06

Hi guys, I need /tmp/ser_fifo to *always* be owned by root:foobar. What's the best way of accomplishing this? I was thinking of chown'ing the fifo at the end of ser's init script. However, I'm not sure when the fifo is created when starting up ser, or whether the fifo is ever recreated (thus requiring a new chown). Any suggestions? I'm all ears! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

2 1

[Users] avp_db_load problem
by Jayesh Nambiar 12 Apr '06

12 Apr '06

hi, It worked after I changed the "$fU" to "$fu". Now it properly does the avp_db_load(). jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

1 0

[Users] avp_db_load problem
by Jayesh Nambiar 11 Apr '06

11 Apr '06

hi, Sorry there was a typo. I get the message "entered condition" in the logs, which indicates it does not do the avp_db_load. It just enters the condition of the uri. thanks, jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

1 0

[Users] avp_db_load problem
by Jayesh Nambiar 11 Apr '06

11 Apr '06

Hi all, I was trying to do the following: Whenever a user presses *86 and then any number, first openser should check in the subscriber table, whether the value for features column is 1. (I've added an extra column features in the table subscriber with datatype int). If the value is 0, he should be replied with the message Forbidden. I tried doing the following: modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=features;table=subscriber") if(uri=~"sip:\*86[0-9]*@") { log(1,"entered condition"); if(avp_db_load("$fU", "$avp(s:features)/$scheme1")) { log(1,"db load succeeded"); if(!avp_check("$avp(s:features)", "eq/1/g")) { sl_send_reply("403", "Forbidden"); exit; }; strip(3); log(1,"stripped"); avp_db_query("insert into usr_preferences values('', '$fU', '', 'callfwd', '', '$ruri', 'CURRENT_TIMESTAMP')"); sl_send_reply("200", "OK"); exit; }; }; But in the logs I only get "condition succeeded" which means it does not do the avp_db_load() properly. Can someone please help me in what should be the exact syntax to load it from the db and then check the value. thanks in advance, jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

1 0

[Users] Random error with NAT (error in 'c field')
by Bruno Machado 11 Apr '06

11 Apr '06

Hi Bogdan Do you want to see all the SIP messages from this call? Is it? Thanks for yor help. --------------------------------- Yahoo! Search Imposto de Renda 2006: o prazo está acabando. Faça já a sua declaração no site da Receita Federal.

2 1

[Users] Forward response
by Benjamin Lawetz 11 Apr '06

11 Apr '06

Hello, Classic situation, overgrown asterisk and need to go openser for load-balancing and failover reasons. We want to do an incremental upgrade to openser, and for the moment we are trying to use openser as a fancy router in front of asterisk. I'm having a little problem with load-balancing the registration. Clients are mostly behind NAT sending registration to openser. I'm using the dispatcher module to load-balance it. The registration is going from the client to the openser, openser is forwarding it to asterisk. Asterisk replies to openser, and that's where I'm stuck. I want openser to forward the "TRYING" and "OK" response (and obviously all other responses) to the client, but can't seem to figure out how. Can anybody point me in the right direction? thanks

2 3

[Users] AVPOPS manipulating header fields
by Martin Kjeldsen 11 Apr '06

11 Apr '06

Hi all I am in the need for a little assistance on how to best use some data taken from a invite request, and then create an appropriate AVP. My idea was to take the from header field, avp_write("$from", "$fromuri"); Then use it to extract the alias which belongs to this user, and create the AVP, avp_db_load("$fromuri","s:pai/$scheme0"); And then incorperate it into another AVP, in order to add a callerid, via P-Asserted-Identity, avp_printf("i:10","<sip:$avp(s:pai)@sip.domain.com>"); remove_hf("P-Asserted-Identity"); #remove org. hdr avp_pushto("$P-Asserted-Identity/request","i:10"); # set new hdr My question is, how do I best remove the port (eg. :5060) indication at the end of the from header field? Because the clients on a private IP range and my alias table does not carry the port numbers (eg. 88888888(a)sip.domain.com ). So I need to manipulate the $fromuri, by removing the port indication, before i do the avp_db_load! I was thinking of using the avp_subst function, but cant quite seem to get it to work. Can anyone help me with a smart way to solve this problem? All help greatly appreciated. Kind regards Martin Kjeldsen

2 1

[Serusers] BYE
by Thomas Britis 11 Apr '06

11 Apr '06

Hi all, Is there a way to finish a call without a BYE? Here is my problem: I have my own billing system and I terminate a established call just when I get a BYE message from client (or gateway). So I was thinking if one could terminate the call without the BYE. Thank you. -- Thomas Storino Britis TCNet Informatica e Telecomunicacoes LTDA

2 1

[Users] Fwd:Invalid table version
by buxalex＠libero.it 11 Apr '06

11 Apr '06

Hello, I tried also to drop and the to install the db, but still I get the same error.... Please anybody can help me? I am using openser version 1.1.0 dev17 tls thanks in advance Alessandro ---------------------------------------------------------------- Hello, running openser with mysql db (through monitor), I retrieve: ACC - initializing Apr 11 18:58:32 dhc27-53-93pc /usr/local/sbin/openser[7806]: register_udomain(): Invalid table version (use openser_mysql.sh reinstall) If I comment the module ACC loading I get the same error for the other modules. I reinstalled, as suggested from the log the db, but still I get the error. in table version I can see that the domain table has version 1 Any idea? This is a test db, do you suggest to erase it and to install it again? thanks a lot Alessandro

2 1

[Serusers] Release 0.9.6: modparam syntax for radius accounting
by Alberto Cruz 11 Apr '06

11 Apr '06

Hi Jan I recently install a new SER server with Release 0.9.6 using the debian distribution and I'm trying to use the radius configuration that I have running at the SER server with Release 0.9.3 but I'm having troubles with the following lines: modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "report_ack", 0) when I test the ser.cfg I got "ERROR: bad config file (4 errors)" I was looking for documentation about acc_radius module but I didn't find documentation about it. Is there someplace where I can find the new syntax for the modparam ? Regards Alberto Cruz

3 5

[Users] Invalid table version
by buxalex＠libero.it 11 Apr '06

11 Apr '06

Hello, running openser with mysql db (through monitor), I retrieve: ACC - initializing Apr 11 18:58:32 dhc27-53-93pc /usr/local/sbin/openser[7806]: register_udomain(): Invalid table version (use openser_mysql.sh reinstall) If I comment the module ACC loading I get the same error for the other modules. I reinstalled, as suggested from the log the db, but still I get the error. in table version I can see that the domain table has version 1 Any idea? This is a test db, do you suggest to erase it and to install it again? thanks a lot Alessandro

1 0

[Serusers] Realtime Panel
by Waldo Rubinstein 11 Apr '06

11 Apr '06

Is there anything like the Flash Operator Panel for Asterisk (http:// www.asternic.org/) available for SER? I have been running my systems with Asterisk and would like to start migrating to SER, but kind of need similar functionality. Thanks, Waldo

1 0

[Serusers] Help:SER-RADIUS-LDAP
by Ivan Turcin 11 Apr '06

11 Apr '06

Hi, I'm using SER 0.9.6 as my SIP proxy, and free RADIUS 1.1.0 for accounting, authorization and authentication. Users are in open LDAP 2.3.20. For connecting to RADIUS I'm using auth_radius module which uses radusclient-ng. Everything works fine when digest is used for authentication and authorization but when I try to use LDAP for authentication and authorization i get this from RADIUS: rlm_ldap: - authorize rlm_ldap: performing user authorization for 201(a)192.168.19.2 radius_xlat: '(uid=201(a)192.168.19.2)' radius_xlat: 'ou=People,dc=sips,dc=tel,dc=fer,dc=hr' rlm_ldap: ldap_get_conn: Checking Id: 0 rlm_ldap: ldap_get_conn: Got Id: 0 rlm_ldap: attempting LDAP reconnection rlm_ldap: (re)connect to 192.168.19.2:389, authentication 0 rlm_ldap: bind as cn=root,dc=sips,dc=tel,dc=fer,dc=hr/topsct to 192.168.19.2:389 rlm_ldap: waiting for bind result ... rlm_ldap: Bind was successful rlm_ldap: performing search in ou=People,dc=sips,dc=tel,dc=fer,dc=hr, with filter (uid=201(a)192.168.19.2) rlm_ldap: checking if remote access for 201(a)192.168.19.2 is allowed by employeeType rlm_ldap: Added password 201 in check items rlm_ldap: looking for check items in directory... rlm_ldap: Adding userPassword as User-Password, value 201 & op=21 rlm_ldap: looking for reply items in directory... rlm_ldap: user 201(a)192.168.19.2 authorized to use remote access rlm_ldap: ldap_release_conn: Release Id: 0 modcall[authorize]: module "ldap" returns ok for request 0 modcall: leaving group authorize (returns ok) for request 0 rad_check_password: Found Auth-Type Ldap auth: type "LDAP" Processing the authenticate section of radiusd.conf modcall: entering group LDAP for request 0 rlm_ldap: - authenticate rlm_ldap: Attribute "User-Password" is required for authentication. modcall[authenticate]: module "ldap" returns invalid for request 0 modcall: leaving group LDAP (returns invalid) for request 0 auth: Failed to validate the user. I think this is the problem: Attribute "User-Password" is required for authentication. In users file I have added:a DFAULT Auth-Type:=LDAP to force using LDAP for authentication and authorization. When i try to connect remotly using radius client from command line, authorization and authentication works fine. When I capture packets when using SER i can't see User-Password attribute. Is there any way to solve this problem. May be to say in RADIUS that some of digest attributes is actually User-Password attribute, or some other module which enables using of RADIUS and LDAP. Thanks in advance. Best regards, -- Ivan Turcin Student at University of Zagreb, Faculty of Electricalengeniring and Computing, Branch of Telecomunications and Informatics Unska 3 HR-10000 Zagreb

2 1

[Users] nat_helper: multiple media IP address in SDP
by Nicolas Olivier 11 Apr '06

11 Apr '06

Hi, I've got a gateway which is only used for rounting and rtp proxying between providers and centrexes. On reply to an INVITE, one of our provider send back a "183 Session Progress". The problem is that in the SDP block, we've got 2 media IP address and rtpproxy only rewrite one. Finally, the provider establish rtp session with our gateway, and our centrex directly with the provider. provider gateway centrex 172.16.0.10 192.168.1.10 192.168.1.20 RTP -------------> RTP ------------> RTP ^-------------------------------------------------| So my questions are, is it possible to have multiple IP address in SDP and if so, how can I tell rtpproxy to rewrite all of them. Coming from provider: SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.10;branch=z9hG4bKdd67.a4cc2c44.0,SIP/2.0/UDP 192.168.1.20:5062;branch=z9hG4bKdd67.08f45a33.0,SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK4af242b7. From: "02" <sip:0143132445@192.168.1.20>;tag=as226ce7b9. To: <sip:0123456789@192.168.1.20:5062>;tag=3123AAA8-20C5. Date: Tue, 11 Apr 2006 09:10:29 GMT. Call-ID: 079ab6663e403ff44a1107e5111b075f(a)192.168.1.20. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow-Events: telephone-event. Contact: <sip:677238#0123456789@172.16.0.10:5060>. Record-Route: <sip:192.168.1.10;ftag=as226ce7b9;lr=on>,<sip:192.168.1.20:5062;ftag=as226ce7b9;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 261. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3448 4768 IN IP4 172.16.0.10. s=SIP Call. c=IN IP4 172.16.0.10. t=0 0. m=audio 18322 RTP/AVP 18 101. c=IN IP4 172.16.0.10. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Forwarded to centrex: SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.20:5062;branch=z9hG4bK43a4.3e96aba3.0,SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3213db83. From: "02" <sip:0143132445@192.168.1.20>;tag=as1a2f900d. To: <sip:0123456789@192.168.1.20:5062>;tag=3121D1B4-1BFD. Date: Tue, 11 Apr 2006 09:08:28 GMT. Call-ID: 08467c5e299ab833106517c63d3edc2e(a)192.168.1.20. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow-Events: telephone-event. Contact: <sip:677238#0123456789@172.16.0.10:5060>. Record-Route: <sip:192.168.1.10;ftag=as1a2f900d;lr=on>,<sip:192.168.1.20:5062;ftag=as1a2f900d;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 277. . v=0. o=CiscoSystemsSIP-GW-UserAgent 565 174 IN IP4 172.16.0.10. s=SIP Call. c=IN IP4 172.16.0.10. t=0 0. m=audio 36296 RTP/AVP 18 101. c=IN IP4 192.168.1.10. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=nortpproxy:yes. openser.cfg (...) onreply_route[1] { if (status =~ "(180)|(183)|2[0-9][0-9]") { fix_nated_contact(); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; } else if (nat_uac_test("1")) { fix_nated_contact(); }; } (...) Best regards, Nicolas Olivier

2 4

[Serusers] Gateway authentication
by Daniel Korndorfer 11 Apr '06

11 Apr '06

Does anybody know how can I send credentials to a gateway? For example: I forward a call using forward(uri:host, uri:port);but my gateway needs authentication. Tks, D.K.

3 3

[Users] Bug in avp_db_load?
by Andreas Granig 11 Apr '06

11 Apr '06

Hi, Is it a bug or my fault? http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_db_load states for the syntax of avp_db_load(source, name): name - which AVPs will be loaded from DB into memory. This can be an avp_name or an avp_alias. It can be empty - which means all AVPs, or you may specify only the type - 'i:' means all AVPs with ID. Now if I try this one (CVS HEAD): avp_db_load("$avp($callee_uuid)","i:"); or this one: avp_db_load("$avp($callee_uuid)",""); with $avp($callee_uuid) being '2', the corresponding DB query *always* looks like this: submit_query(): select value,attribute,type from usr_preferences where uuid='2' AND attribute='0' which doesn't return a result. I'd rather expect the query being "select value,attribute,type from usr_preferences where uuid='2' AND type=2" and "select value,attribute,type from usr_preferences where uuid='2'" for my two statements. The queried table looks like this: mysql root@openser> select * from usr_preferences where uuid='2'\G *************************** 1. row *************************** uuid: 2 username: 012345 domain: my.domain attribute: 100 type: 2 value: 01 modified: 2006-04-11 11:21:22 Thanks, Andy

2 2

[Users] load from db table
by D'Addelfio Davide 11 Apr '06

11 Apr '06

Hi Bogdan, i setup my openser.cfg to store SIP messages into acc tables of mysql, using extra accounting to store also the body part of the message. Now i need that openser read into that db's table, in particular in some rows of db. If I use avp_db_load it works only over usr_preference table, instead I want it looks into acc table. How can I do? Thanks for help Davide

3 2

[Serusers] SER and Radiusclient-ng on 64bit Linux OS
by Alan 11 Apr '06

11 Apr '06

I've compiled and installed the sources successfully on a Sun Mircosystems 20z with Red Hat Enterprise v3 running the OS. When I try to use radius it always fails to log the user in and authenticate with radius. If I run SER on a 32 bit system with the same configurations as the 64 bit system it works fine. After reading some documentation, I caught reference in regards to radiusclient-ng only supported on 32 bit systems. Is this true? The only thing that doesn't work is when a user tries to authenticate through radius the authentication always fails on the 64 bit OS. Am I missing any pre-configuration options prior to compiling on a 64 bit OS? Please help. ~Alan Error message received from debug: pre_auth(): Credentials with given realm not found

2 1

[Users] openser-1.0.1-rhel3.0.i386.rpm
by Nasser Dawoud /MEM/PRO 11 Apr '06

11 Apr '06

Hi everyone, I have just installed openser, openser-1.0.1-rhel3.0.i386.rpm, and I have a couple of questions: 1) is this version of OpenSer fully compliant with the Presence portion of the SIP-RFC? 2) I've noticed that presence is buggy, it does not always work correctly; that is to say, it does not represent the correct presence of the respective user in the roster. 3) I would like to use OpenSer and TLS, does that mean I only replace the libraries, *.so files, or do I have to replace everything?

1 0

[Users] callid field in MYSQL
by Shane Burrell 11 Apr '06

11 Apr '06

What should be in this field? A unique ID or a the caller id? TIA Shane

2 1

[Users] Random error with NAT (error in 'c field')
by Bruno Machado 11 Apr '06

11 Apr '06

Hi fellows Our system is showing an anormal behavior. Sometimes, our clients devices behind NAT cannot obtain successfuls call to other device. Take a look at the SDP headers below. These 2 dumps were made between SIP Proxy and client (at private IP 192.168.0.13). Notice that the 'c field' is differente between them. The first SDP has the public address from RTPProxy (X.Y.Z.A). This call is a example of a perfect situation. The SDP below that shows the 'c field' with a private address. When the call is established, the RTP from my phone is sent to this private IP, 192.168.0.13, in our LAN. All the sinalization is sent among the 2 devices and SIP Proxy normally. When the 'openser' process is restarted, the calls flows without any problems. Somebody knows what can be happen? Thanks in advance Bruno Machado .................. INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 402 P-hint: usrloc applied P-hint: nat-p2p v=0 o=11000600 8000 8002 IN IP4 192.168.0.13 s=SIP Call c=IN IP4 X.Y.Z.A..t=0 0 m=audio 56304 RTP/AVP 18 4 0 8 2 99 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 ................... INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 381 P-hint: usrloc applied P-hint: nat-p2p v=0 o=11000600 8000 8002 IN IP4 192.168.0.13 s=SIPCall c=IN IP4 192.168.0.13 t=0 0 m=audio 5004 RTP/AVP 18 4 0 8 2 99 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000..a=rtpmap:8 ....................... __________________________________________________ Faça ligações para outros computadores com o novo Yahoo! Messenger http://br.beta.messenger.yahoo.com/

2 1

[Users] Media
by Rafael Neves Harff 11 Apr '06

11 Apr '06

Hi! I've finished installing Openser on my Slackware box. I can athenticate and make SIP calls. I need to make media traffic pass through the server. Is it possible using Openser? How? If Openser doesn't implement this feature is there a 3rd software I can use to do this using Openser? Thanks, Rafael Neves Harff from Brazil

2 1

[Users] uuid-based setup
by Andreas Granig 11 Apr '06

11 Apr '06

Hi again, Just another question: If I'd like to switch from a username/domain-based setup to an uuid-based setup, what exactly would I have to consider? As far as I have found in the docs (or better: what I assume from what I've found in the docs), I'd have to load the uuid from subscriber like avp_db_query("select uuid from subscriber where username='$au' and domain='$ar'", "$avp(i:500)"); And then I can use i:500 as uuid for loading other avps? Or is there an implicit way to get the uuid, e.g. during www/proxy_authorize? I think I have some troubles to get the link between username/domain and uuid... Thanks, Andy

2 2

[Users] Reading table from db
by D'Addelfio Davide 11 Apr '06

11 Apr '06

Hi all, i have configured my openser to store SIP messages into acc tables of mysql, now i need that openser read into that db's table. How can I do? Thanks

1 0

[Users] Allow only TLS connections
by Thorsten.Haupt＠t-systems.com 11 Apr '06

11 Apr '06

Hello, I use OpenSER in a testing environment for VoIP security. My clients connect via TLS. If I deactivate UDP/5060 on the server, it doesn't work correct. Some Clients can't connect and others can't establish calls. I read in another thread, that UDP is mandatory for SIP and that the server need it. But how can I prevent users from connecting via UDP and force them to use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But is this the correct way? Are there any parameters server-side to force users to connect via TLS? Thanks for response. Torsten

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[Users] OpenSER using TLS / TCP
by Thorsten.Haupt＠t-systems.com 11 Apr '06

11 Apr '06

> I think snoms are not capable of receiving incoming tls connection. And MiniSIP ... ? > Thus, you have to keep the snom-openser tls connection open ... a way to do it is to force the snom phone the register > before the tls timeout ... i think that the timeout is around 2 minutes ... Yes, this is one solution. My telephones (Snoms and MiniSIP) register now every minute. But is there no other solution, server-side? What about the parameters "tls_handshake_timeout", "tls_send_timeout" or "tcp_connect_timeout"? I tried this parameters, but nothing changed. Torsten On 4/10/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > > > Hello, > > I have a problem with OpenSER using TLS / TCP, Clients are MiniSIP and > Snom 360 phones. > I start the server, the phones connect to it using TCP or TLS over TCP. > Calls between all phones are possible. > But after some minutes, I think it is the timeout of TLS or TCP, no > calls are possible any longer. > OpenSER shows the following error message: > > 8(31121) ERROR: tcp_blocking_connect: SO_ERROR (111) Connection > refused > 8(31121) ERROR: tcpconn_connect: tcp_blocking_connect failed > 8(31121) ERROR: tcp_send: connect failed > 8(31121) msg_send: ERROR: tcp_send failed > 8(31121) ERROR:tm:t_forward_nonack: sending request failed > > Has anyone a solution for this problem? > Thanks. > > Torsten > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Method BYE and loose_routed function
by buxalex＠libero.it 11 Apr '06

11 Apr '06

Hello Klaus, you were right, my BYE is loose_routed. In fact, following your suggestion, I've got the table updated. My question now is: how to know, when a call is loose_routed? I mean, in both cases now the configuration update the ACC table. should I create a if case or something like that? Thanks in advance for your help Alessandro ---------- Initial Header ----------- >From : "Klaus Darilion" klaus.mailinglists(a)pernau.at To : "buxalex(a)libero.it" buxalex(a)libero.it Cc : "users" users(a)openser.org Date : Mon, 10 Apr 2006 16:45:49 +0200 Subject : Re: [Users] new module: siptrace > Hi Alessandro! > > Please do not reply to emails when creating a new thread. Use "new > email" and choose the proper subject. Otherwise the "thread view" > feature of some email clients don't work anymore. > > Regarding your problem. May it be that the BYE is loose_routed and thus > does not hit the setflag()? > > Try setflag(2) in loose_route block. > > regards > klaus > > > buxalex(a)libero.it wrote: > > Hello to everybody, > > > > I am running openser with the module ACC and I get the db correctly updated (table ACC) for the method INVITE and ACK; But I cannot retrieve the method BYE, I am sure I have made mistake in openser.cfg, anybody can give me any help? below my openser.cfg: > > > > #----------------------------------- > > # ACC parameters > > #----------------------------------- > > > > modparam("acc", "db_url", "mysql://openser:openser@localhost/openser") > > modparam("acc", "failed_transaction_flag", 4) > > modparam("acc", "report_cancels", 1) > > modparam("acc", "db_flag", 2) > > modparam("acc", "db_missed_flag",3 ) > > .... > > > > if (method=="BYE") { > > setflag(2); > > t_relay(); > > return; > > }; > > > > > > if (method=="INVITE") { > > setflag(2); > > record_route(); > > }; > > lookup("aliases"); > > if (!uri==myself) { > > append_hf("P-hint: outbound alias\r\n"); > > route(1); > > }; > > > > thank you very much > > Alessandro > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

RE: [Serusers] How can i setup SER on Cygwin?
by Garima Mishra 11 Apr '06

11 Apr '06

hi brett, Actually, i want to setup SER on windows, so i tried using Cygwin. But if anyone of you have done this through some other means then please tell me. Garima -----Original Message----- From: Brett N [mailto:brettlist@nemeroff.com] Sent: Friday, April 07, 2006 8:25 PM To: Garima Mishra Cc: 'serusers(a)lists.iptel.org' Subject: Re: [Serusers] How can i setup SER on Cygwin? Your kidding right? Why are you using Cygwin? -Brett On Fri, April 7, 2006 2:11 am, Garima Mishra wrote: > > i am trying to setup SER on cygwin but during compilation i am getting > errors like > "Makefile.rules:80: qvalue.d: No such file or directory" > same error for many files... > > can anyone suggest me how can i setup SER on cygwin? > > Thanks, > Garima > > Garima Mishra > > Software Engineer > <http://www.openeratech.com/> <http://www.openeratech.com/> > <http://www.openeratech.com/> > > Openera Technologies Pvt Ltd. > 97, Block 5 > Koramangala Indl. Estate > <?xml:namespace prefix = st1 ns = > "urn:schemas-microsoft-com:office:smarttags" />Bangalore - 560 095. India > > Tel: +91 80 2552 1937 > Fax: +91 80 2552 1164 > <http://www.openeratech.com/> www.openeratech.com > > E-mail: <mailto:garima@openeratech.com> garima(a)openeratech.com > > > > This message (including any attachments) is a confidential and Privileged > communication to the intended addressee. If you are not the Intended > addressee, you have received this message in error. In that case, please > permanently delete this message and notify the sender so that we can avoid > this inconvenience in the future. Thank you. > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Users] extra accounting usage
by D'Addelfio Davide 11 Apr '06

11 Apr '06

Hi to all, It seems I fix the problem, setup config file in this way and he write in msg_body column the SDP part of the message loadmodule "/usr/lib/openser/modules/acc.so" modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "db_table_acc", "msg_body") modparam("acc","db_extra", "msg_body=$rb") if (method=="INVITE") acc_db_request("Prova", "acc"); thanks davide

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RE: [Serusers] variables which can be used when writing log
by İlker Aktuna (Koç.net) 11 Apr '06

11 Apr '06

Thanks, That was really what I'm looking for. Sorry, I didn't know that this was in the source. Thanks a lot ! -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Michal Matyska Sent: Monday, April 10, 2006 6:00 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] variables which can be used when writing log If you wan to use xlog just look into xlog's source. You'll find long switch statement there listing all the available dual char codes. Or see "grep -A 1 case xl_lib.c" result and use pattern: %br - branch %bR - branches, and so on %C. are color switches. case 'b': case 'r': e->itf = xl_get_branch; case 'R': e->itf = xl_get_branches; case 'c': case 'i': e->itf = xl_get_callid; case 's': e->itf = xl_get_cseq; case 't': e->itf = xl_get_contact; case 'C': case 'x': case 's': case 'r': case 'g': case 'y': case 'b': case 'p': case 'c': case 'w': case 'S': case 'R': case 'G': case 'Y': case 'B': case 'P': case 'C': case 'W': case 'x': case 's': case 'r': case 'g': case 'y': case 'b': case 'p': case 'c': case 'w': case 'd': case 's': e->itf = xl_get_dset; case 'f': case 'u': e->itf = xl_get_from; case 't': e->itf = xl_get_from_tag; case 'H': case 'n': e->hindex = XL_HOST_NAME; case 'd': e->hindex = XL_HOST_DOMAIN; case 'f': e->hindex = XL_HOST_FULL; case 'i': e->hindex = XL_HOST_IPADDR; case 'm': case 'b': e->itf = xl_get_msg_buf; case 'f': e->itf = xl_get_flags; case 'i': e->itf = xl_get_msgid; case 'l': e->itf = xl_get_msg_len; case 'x': e->itf = xl_get_msgid; case 'n': case 'h': e->itf = xl_get_nexthop; case 'p': case 'p': e->itf = xl_get_pid; case 'x': e->itf = xl_get_pid; case 'r': case 'm': e->itf = xl_get_method; case 'u': e->itf = xl_get_ruri; case 's': e->itf = xl_get_status; case 'r': e->itf = xl_get_reason; case 't': e->itf = xl_get_refer_to; case 'R': case 'i': e->itf = xl_get_rcvip; case 'p': e->itf = xl_get_rcvport; case 's': case 'i': e->itf = xl_get_srcip; case 'p': e->itf = xl_get_srcport; case 't': case 'u': e->itf = xl_get_to; case 't': e->itf = xl_get_to_tag; case 'T': case 's': e->itf = xl_get_times; case 'f': e->itf = xl_get_timef; case 'x': e->itf = xl_get_times; case 'u': case 'a': e->itf = xl_get_useragent; case 'q': e->itf = xl_get_unique; Michal On Mon, Apr 10, 2006 at 05:40:15PM +0300, İlker Aktuna (Koç.net) wrote: > Hi, > > This surely helps a lot but XLOG uses 2 character variable names to write in logging. I need those variable names and this document does not include them as far as I can see. > > Thanks, > ilker _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers <http://387555.sigclick.mailinfo.com/sigclick/0B090704/04024D00/0B094C03/181…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

RE: [Users] Force RTP stream to go through mediaproxy
by Sam Lee 11 Apr '06

11 Apr '06

any help ? ________________________________ From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Sam Lee Sent: Sunday, April 09, 2006 12:48 PM To: users(a)openser.org Subject: [Users] Force RTP stream to go through mediaproxy Hi all, I got the openser and mediaproxy up working and fine. The mediaproxy is able to work with certain NATED configurations, but not all. When i tried to check what is the problem , those NATED configurations that were not working was found not to have made use of the mediaproxy, which causes the problem. What i am trying to do now, but without much success, is to force all the RTP media to go through the mediaproxy. Is there any way i can force all the RTP media to go through the mediaproxy without all the client_nat_test stuff...? Any hints will be much appreciated ! Regards, Sam

1 0

[Serusers] Re: Serusers Digest, Vol 36, Issue 8
by xirak＠netscape.net 11 Apr '06

11 Apr '06

-----Original Message----- Brett, I have looked at your post and was wondering if this is something you're working on? What's the status of the module? Zak ------------------------------ Message: 20 Date: Sat, 8 Apr 2006 01:54:02 -0500 (CDT) From: "Brett N" <brettlist(a)nemeroff.com> Subject: Re: [Serusers] Limit number of calls To: zakforever(a)netscape.net Cc: serusers(a)lists.iptel.org Message-ID: <64448.24.206.109.215.1144479242.squirrel(a)mail.nemeroff.com> Content-Type: text/plain;charset=iso-8859-1 Search the list archives, I posted regarding this about a week ago.. On Fri, April 7, 2006 3:46 pm, zakforever(a)netscape.net wrote: > Hi All, > > Is there a way in SER that would limit the number of calls that any > user can make at any one time to just one call ? > > Zak > ___________________________________________________ > Try the New Netscape Mail Today! > Virtually Spam-Free | More Storage | Import Your Contact List > http://mail.netscape.com > > _______________________________________________ > 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 End of Serusers Digest, Vol 36, Issue 8 *************************************** ___________________________________________________ Try the New Netscape Mail Today! Virtually Spam-Free | More Storage | Import Your Contact List http://mail.netscape.com

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[Serusers] Checking spoofed From headers
by sip 10 Apr '06

10 Apr '06

Is there a way anyone can readily think of to check to see if someone using our open proxy is calling through with a From header that attempts to fool the recipient into thinking the call is validly from one of our users? Scenario is this... While looking at the logs this morning, I noticed someone was calling a SIPPhone user through our proxy with a From: address that LOOKED like it was a user of ours, but using a username that doesn't actually exist. I'm wondering if there's anyway to check if someone is calling through us with a From: address that looks like one of our users, but isn't. N.

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Re: [Users] Recursive routes and retcode
by Andreas Granig 10 Apr '06

10 Apr '06

Hi Klaus, > What is your goal? Do you want to strip the extension till you get the > "kopfnummer" (sorry, I do not know the english term)? Maybe "base number" or something like that? Yes, that's exactly what I'm trying to find out. We've used a self-made lookup function until now, but in the course of the switch from SER to OpenSER I try to eliminate as much self-written stuff as possible for easier maintenance. I'm aware of the performance impact, but I'm currently just playing around with some stuff to see what's possible and maybe usable... > I know other which solve this problem by using a private ENUM tree with > wildcard NAPTRs for the extensions which point to the "main URI". We use a related approach in the e164.info tree. I'm quite curious about the performance of the different possibilities, so I find it rather interesting to compare these things in a test-bed... Andy

1 0

[Fwd: Re: [Users] Recursive routes and retcode]
by Klaus Darilion 10 Apr '06

10 Apr '06

forgotten to CC the list. regards klaus ---------------------------- Original Message ---------------------------- Subject: Re: [Users] Recursive routes and retcode From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> Date: Mon, April 10, 2006 23:36 To: "Andreas Granig" <andreas.granig(a)inode.info> -------------------------------------------------------------------------- Hi Andy! What is your goal? Do you want to strip the extension till you get the "kopfnummer" (sorry, I do not know the english term)? Maybe a special lookup function might be useful which does some kind of longest match matching (like the SQl query in the LCR module). I know other which solve this problem by using a private ENUM tree with wildcard NAPTRs for the extensions which point to the "main URI". regards klaus On Mon, April 10, 2006 18:44, Andreas Granig said: > Hi, > > I'd like to recursively strip off digits from the tail of a username (a numeric number in fact) and do a lookup in usrloc until an aor is found or the number underruns a lower bound, like: > > route[2] { > if(lookup("location")) return(2); > if(!uri =~ "^sip:[0-9]{6}[0-9]*@") return(3); > strip_tail(1); > route(2); > } > > route[3] { > # check if numeric, then: > route(2); > if(retcode==1) {...} > else if(retcode==2) {...} > else if(retcode == 3) {...} > # ... > } > > The thing is that it obviously doesn't work (retcode is 1 as soon as one recursion happens) because I can't tell the recursive call to properly return the retcode like "return(route(2))" or "return($?)" or something like that. Any other ideas except using flags or is that the only way to go? > > Thanks, > Andy > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Users] Recursive routes and retcode
by Andreas Granig 10 Apr '06

10 Apr '06

Hi, I'd like to recursively strip off digits from the tail of a username (a numeric number in fact) and do a lookup in usrloc until an aor is found or the number underruns a lower bound, like: route[2] { if(lookup("location")) return(2); if(!uri =~ "^sip:[0-9]{6}[0-9]*@") return(3); strip_tail(1); route(2); } route[3] { # check if numeric, then: route(2); if(retcode==1) {...} else if(retcode==2) {...} else if(retcode == 3) {...} # ... } The thing is that it obviously doesn't work (retcode is 1 as soon as one recursion happens) because I can't tell the recursive call to properly return the retcode like "return(route(2))" or "return($?)" or something like that. Any other ideas except using flags or is that the only way to go? Thanks, Andy

3 3

[Users] OpenSer 1.0.1 with MAXTNT PSTN and rtpproxy sample
by Shane Burrell 10 Apr '06

10 Apr '06

Anyone currently doing this care to share some insight on getting it all working. I seem to have trouble with the rtpproxy section handling NAT correctly. The TNT is taking calls and MYSQL is working. TIA Shane

1 0

RE: [Serusers] serctl - serweb
by Gould, Aaron 10 Apr '06

10 Apr '06

i downloaded a couple xxx.gz.tar files and opened them up. what are the .cfg files used for? like what is the "hello-world.5.0.cfg" supposed to be used for? aaron ________________________________ From: Gould, Aaron Sent: Mon 4/10/2006 8:52 AM To: samuel Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] serctl - serweb thanks samuel good, then i'm not going to mess with setting up apache for serweb if i can do all that is neeed from serctl. you probably don't get what i'm saying because i don't get what i'm saying. ;) get it? :) i'm still learning about sip. ...and trying to set it up as i go. i'll take your advice and read the getting started docs at www.onsip.org <http://www.onsip.org> i think what kicked off the 70 hop decrement to 0 (zero) was the dns entry i made for _sip._udp.lab.com and put in that entry the host of the ser box as the host providing the service. was this wrong? i might have the cart before the horse in some of this madness. i'm still trying to put the puzzle peices together and understand how it all works. now that i've removed the _sip._udp.lab.com dns entry , i know get from the sre box a response from the UA softphone register message i i now get from the ser box a 478 error Unresolvable destination aaron ________________________________ From: samuel [mailto:samu60@gmail.com] Sent: Mon 4/10/2006 8:36 AM To: Gould, Aaron Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] serctl - serweb 2006/4/10, Gould, Aaron <aaron.gould(a)ngc.com>: > i'm still trying to get a UA (softphone to "REGISTER") with my ser box (solaris 10) > > do serctl and serweb serve the same purpose? No. serweb is a web interface for the user management. serctl is the tools itself to manage users. > > do i really need to setup serweb? You can do everything via command line with serctl > > can i do everything via serctl? > > also, is there such a thing as auto-registration with UA's to the registrar of the ser box? > I don't really get what you are trying to accomplish here... > if not, must i do a "serctl add" to add a user first? What #serctl add does is to add user's profile to your system so they can later use the profile data to register their phones into your system. > > something I'm seeing is, my softphone sends a "REGISTER" message to the ser box. the ser box appears to send 70 (i think 70 because of the default max-forwards value) dns request to the dns server, then the ser box sends a "483 Too many Hops" message back to the latop UA softphone. any ideas why this is happening? Probably your config file is wrong. I recommend to go to www.onsip.org and read the getting started document that includes a detailled description of several configurations. Probably, the Too Many Hops comes from a paquet being forwarded to yourself (to the same SER) until the hop count decresases too low. > > > thanks > You'r wellcome, Samuel. > aaron > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Users] OpenSER using TLS / TCP
by Thorsten.Haupt＠t-systems.com 10 Apr '06

10 Apr '06

Hello, I have a problem with OpenSER using TLS / TCP, Clients are MiniSIP and Snom 360 phones. I start the server, the phones connect to it using TCP or TLS over TCP. Calls between all phones are possible. But after some minutes, I think it is the timeout of TLS or TCP, no calls are possible any longer. OpenSER shows the following error message: 8(31121) ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused 8(31121) ERROR: tcpconn_connect: tcp_blocking_connect failed 8(31121) ERROR: tcp_send: connect failed 8(31121) msg_send: ERROR: tcp_send failed 8(31121) ERROR:tm:t_forward_nonack: sending request failed Has anyone a solution for this problem? Thanks. Torsten

2 1

[Users] extra accounting usage
by D'Addelfio Davide 10 Apr '06

10 Apr '06

Does anyone help me to understand how set-up config file to work with extra accounting? I need to let it write into db the body part of the INVITE message but I'm not sure to understand at all the docs... modparam("acc", "db_url", "mysql://user:password@localhost/openser") modparam("acc", "db_extra", "ct=$rb") I think I miss something, cause it doesn't work...!!! Please help me!!! Davide

1 0

RE: [Serusers] serctl - serweb
by Gould, Aaron 10 Apr '06

10 Apr '06

thanks samuel good, then i'm not going to mess with setting up apache for serweb if i can do all that is neeed from serctl. you probably don't get what i'm saying because i don't get what i'm saying. ;) get it? :) i'm still learning about sip. ...and trying to set it up as i go. i'll take your advice and read the getting started docs at www.onsip.org <http://www.onsip.org> i think what kicked off the 70 hop decrement to 0 (zero) was the dns entry i made for _sip._udp.lab.com and put in that entry the host of the ser box as the host providing the service. was this wrong? i might have the cart before the horse in some of this madness. i'm still trying to put the puzzle peices together and understand how it all works. now that i've removed the _sip._udp.lab.com dns entry , i know get from the sre box a response from the UA softphone register message i i now get from the ser box a 478 error Unresolvable destination aaron ________________________________ From: samuel [mailto:samu60@gmail.com] Sent: Mon 4/10/2006 8:36 AM To: Gould, Aaron Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] serctl - serweb 2006/4/10, Gould, Aaron <aaron.gould(a)ngc.com>: > i'm still trying to get a UA (softphone to "REGISTER") with my ser box (solaris 10) > > do serctl and serweb serve the same purpose? No. serweb is a web interface for the user management. serctl is the tools itself to manage users. > > do i really need to setup serweb? You can do everything via command line with serctl > > can i do everything via serctl? > > also, is there such a thing as auto-registration with UA's to the registrar of the ser box? > I don't really get what you are trying to accomplish here... > if not, must i do a "serctl add" to add a user first? What #serctl add does is to add user's profile to your system so they can later use the profile data to register their phones into your system. > > something I'm seeing is, my softphone sends a "REGISTER" message to the ser box. the ser box appears to send 70 (i think 70 because of the default max-forwards value) dns request to the dns server, then the ser box sends a "483 Too many Hops" message back to the latop UA softphone. any ideas why this is happening? Probably your config file is wrong. I recommend to go to www.onsip.org and read the getting started document that includes a detailled description of several configurations. Probably, the Too Many Hops comes from a paquet being forwarded to yourself (to the same SER) until the hop count decresases too low. > > > thanks > You'r wellcome, Samuel. > aaron > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] serctl - serweb
by Gould, Aaron 10 Apr '06

10 Apr '06

i'm still trying to get a UA (softphone to "REGISTER") with my ser box (solaris 10) do serctl and serweb serve the same purpose? do i really need to setup serweb? can i do everything via serctl? also, is there such a thing as auto-registration with UA's to the registrar of the ser box? if not, must i do a "serctl add" to add a user first? something I'm seeing is, my softphone sends a "REGISTER" message to the ser box. the ser box appears to send 70 (i think 70 because of the default max-forwards value) dns request to the dns server, then the ser box sends a "483 Too many Hops" message back to the latop UA softphone. any ideas why this is happening? thanks aaron

2 1

[Serusers] Textops module
by Ladislav Andel 10 Apr '06

10 Apr '06

Hi list, can I change, add or remove SDP lines in with textops module? I'm particularly looking for adding a direction:passive argument to SDP body. Is there any other way to do it? Thank you. Ladislav

2 2

RE: [Serusers] variables which can be used when writing log
by İlker Aktuna (Koç.net) 10 Apr '06

10 Apr '06

Hi, This surely helps a lot but XLOG uses 2 character variable names to write in logging. I need those variable names and this document does not include them as far as I can see. Thanks, ilker ________________________________ From: PREVOT Paul [mailto:prevot@enseirb.fr] Sent: Monday, April 10, 2006 4:36 PM To: İlker Aktuna (Koç.net); 'Ladislav Andel' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] variables which can be used when writing log Maybe it could help you http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook Regards, Paul ________________________________ De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Ilker Aktuna (Koç.net) Envoyé : lundi 10 avril 2006 14:31 À : Ladislav Andel Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] variables which can be used when writing log Hi, Not particularly some variables but anything that would hep me find the problem... In fact, it would be best to learn all of them. Is there a table that shows all ? Maybe in the source code ? (where ?) Thanks, ilker -----Original Message----- From: Ladislav Andel [mailto:ladia6@centrum.cz] Sent: Monday, April 10, 2006 4:31 PM To: İlker Aktuna (Koç.net); serusers(a)lists.iptel.org Subject: Re: [Serusers] variables which can be used when writing log Can you tell me what variables you are particularly looking for? Ladislav İlker Aktuna (Koç.net) wrote: > > > Hi, > > Thanks for your reply. I know the use of XLOG command but how can I > learn other variables ? > Thanks, > ilker > > -----Original Message----- > From: Ladislav Andel [mailto:ladia6@centrum.cz] > Sent: Monday, April 10, 2006 10:22 AM > To: İlker Aktuna (Koç.net) > Subject: Re: [Serusers] variables which can be used when writing log > > Look at XLOG module .. it is very nice tool for logging SIP message > flow through ser.cfg. > > for example > xlog("L_ERR"," Request Method is <%rm>"); an other useful variables > > Ladislav > > İlker Aktuna (Koç.net) wrote: > > Hi, > > > > I need to learn what variables I can use when writing log, for > > debugging purposes. > > Can anyone tell me variables that are avilable in logging ? > > Especially a variable for method name and sip message direction > > would be very useful. > > > > Thanks, > > ilker > > > > -------------------------------------------------------------------- > > -- > > -- > > *From:* serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] > > *On Behalf Of *İlker Aktuna (Koç.net) > > *Sent:* Friday, April 07, 2006 3:36 PM > > *To:* serusers(a)lists.iptel.org > > *Subject:* [Serusers] variables which can be used when writing log > > > > How can I write the uri and myself variables to log file ? > > I know that %tu and %fu are used for to uri and from uri but what > > other variables can I use when writing log ? > > I need to see the "myself" variable in the log because a uri==myself > > statement does not match where it should. > > > > Thanks, > > ilker > > C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, > > > > I need to learn what variables I can use when writing log, for > > debugging purposes. > > Can anyone tell me variables that are avilable in logging ? > > Especially a variable for method name and sip message direction > > would be very useful. > > _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________ <http://387555.sigclick.mailinfo.com/sigclick/010A0504/030C4E03/01004E03/222…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

Re: [Users] new module: siptrace
by buxalex＠libero.it 10 Apr '06

10 Apr '06

Hello to everybody, I am running openser with the module ACC and I get the db correctly updated (table ACC) for the method INVITE and ACK; But I cannot retrieve the method BYE, I am sure I have made mistake in openser.cfg, anybody can give me any help? below my openser.cfg: #----------------------------------- # ACC parameters #----------------------------------- modparam("acc", "db_url", "mysql://openser:openser@localhost/openser") modparam("acc", "failed_transaction_flag", 4) modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag",3 ) .... if (method=="BYE") { setflag(2); t_relay(); return; }; if (method=="INVITE") { setflag(2); record_route(); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; thank you very much Alessandro

2 1

RE: [Serusers] variables which can be used when writing log
by İlker Aktuna (Koç.net) 10 Apr '06

10 Apr '06

Hi, Not particularly some variables but anything that would hep me find the problem... In fact, it would be best to learn all of them. Is there a table that shows all ? Maybe in the source code ? (where ?) Thanks, ilker -----Original Message----- From: Ladislav Andel [mailto:ladia6@centrum.cz] Sent: Monday, April 10, 2006 4:31 PM To: İlker Aktuna (Koç.net); serusers(a)lists.iptel.org Subject: Re: [Serusers] variables which can be used when writing log Can you tell me what variables you are particularly looking for? Ladislav İlker Aktuna (Koç.net) wrote: > > > Hi, > > Thanks for your reply. I know the use of XLOG command but how can I > learn other variables ? > Thanks, > ilker > > -----Original Message----- > From: Ladislav Andel [mailto:ladia6@centrum.cz] > Sent: Monday, April 10, 2006 10:22 AM > To: İlker Aktuna (Koç.net) > Subject: Re: [Serusers] variables which can be used when writing log > > Look at XLOG module .. it is very nice tool for logging SIP message > flow through ser.cfg. > > for example > xlog("L_ERR"," Request Method is <%rm>"); an other useful variables > > Ladislav > > İlker Aktuna (Koç.net) wrote: > > Hi, > > > > I need to learn what variables I can use when writing log, for > > debugging purposes. > > Can anyone tell me variables that are avilable in logging ? > > Especially a variable for method name and sip message direction > > would be very useful. > > > > Thanks, > > ilker > > > > -------------------------------------------------------------------- > > -- > > -- > > *From:* serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] > > *On Behalf Of *İlker Aktuna (Koç.net) > > *Sent:* Friday, April 07, 2006 3:36 PM > > *To:* serusers(a)lists.iptel.org > > *Subject:* [Serusers] variables which can be used when writing log > > > > How can I write the uri and myself variables to log file ? > > I know that %tu and %fu are used for to uri and from uri but what > > other variables can I use when writing log ? > > I need to see the "myself" variable in the log because a uri==myself > > statement does not match where it should. > > > > Thanks, > > ilker > > C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, > > > > I need to learn what variables I can use when writing log, for > > debugging purposes. > > Can anyone tell me variables that are avilable in logging ? > > Especially a variable for method name and sip message direction > > would be very useful. > > <http://387555.sigclick.mailinfo.com/sigclick/07090404/000D4D00/06064F03/141…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

Re: [Serusers] variables which can be used when writing log
by Ladislav Andel 10 Apr '06

10 Apr '06

Can you tell me what variables you are particularly looking for? Ladislav İlker Aktuna (Koç.net) wrote: > > > Hi, > > Thanks for your reply. I know the use of XLOG command but how can I > learn other variables ? > Thanks, > ilker > > -----Original Message----- > From: Ladislav Andel [mailto:ladia6@centrum.cz] > Sent: Monday, April 10, 2006 10:22 AM > To: İlker Aktuna (Koç.net) > Subject: Re: [Serusers] variables which can be used when writing log > > Look at XLOG module .. it is very nice tool for logging SIP message > flow through ser.cfg. > > for example > xlog("L_ERR"," Request Method is <%rm>"); an other useful variables > > Ladislav > > İlker Aktuna (Koç.net) wrote: > > Hi, > > > > I need to learn what variables I can use when writing log, for > > debugging purposes. > > Can anyone tell me variables that are avilable in logging ? > > Especially a variable for method name and sip message direction would > > be very useful. > > > > Thanks, > > ilker > > > > ---------------------------------------------------------------------- > > -- > > *From:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > > *On Behalf Of *İlker Aktuna (Koç.net) > > *Sent:* Friday, April 07, 2006 3:36 PM > > *To:* serusers(a)lists.iptel.org > > *Subject:* [Serusers] variables which can be used when writing log > > > > How can I write the uri and myself variables to log file ? > > I know that %tu and %fu are used for to uri and from uri but what > > other variables can I use when writing log ? > > I need to see the "myself" variable in the log because a uri==myself > > statement does not match where it should. > > > > Thanks, > > ilker > > C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, > > > > I need to learn what variables I can use when writing log, for > > debugging purposes. > > Can anyone tell me variables that are avilable in logging ? > > Especially a variable for method name and sip message direction would > > be very useful. > >

1 0

R: [Users] Parsing SDP messages
by D'Addelfio Davide 10 Apr '06

10 Apr '06

Hi Chris, thanks for reply. Yes, should be a good idea but the sip_trace module is available with 1.1.x openser version, I've installed the 1.0.x debian version that works fine...any chance to have the new debian version? I've tried to get it, searching with google but still got none... Any thought? Davide -----Messaggio originale----- Da: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per conto di Christoph Fürstaller Inviato: lunedì 10 aprile 2006 10.33 A: users(a)openser.org Oggetto: Re: [Users] Prasing SDP messages -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Davide, I also need the SDP messages to extract the keys for SRTP. I tested this with the new siptrace module. With this module I save every SIP INVITE message into DB. So I get the SDP message as well. Afterwards I can get the SDP out of the DB. chris... D'Addelfio Davide wrote: > > Hello, > > i need to work over SDP messages extracting from the SIP requests that > arrive in my openser. Can it do something like parsing into database of > these SDP messages? > > Thanks in advance for reply > Davide > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEOhhPR0exH8dhr/YRAt/YAJ0UtBqxQdGvPC1gv7Trgo75TsxGsQCeIRlN qLfMxdsjR+7Ne3JnDvD5GjY= =jTN6 -----END PGP SIGNATURE----- _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] new module: siptrace
by Daniel-Constantin Mierla 10 Apr '06

10 Apr '06

Hello, a new module was committed into the cvs trunk: siptrace. It allows to store SIP messages processed by OpenSER in a database. The messages are stored just before being forwarded, so the changes made during script processing are reflected as well as the destination IP. The tracing can be turned on/off via fifo command. It uses flags to mark messages to be traced, therefore, filters for groups of users or other criteria can be configured in script. The module gives the possibility to send the traced message to a configurable address. The siptrace module should help companies to deal with some law requirements related to legal interception and calls history. For more info, please see: http://openser.org/docs/modules/1.1.x/siptrace.html Cheers, Daniel

4 5

[Users] Named flags
by Andreas Granig 10 Apr '06

10 Apr '06

Hi, Hasn't there been introduced a new feature for "named" flags some time ago? Like "setflag(my_flag)" or something like this? I think so, but can't find anything about that anymore... Thanks, Andy

2 1

[Serusers] Ser with mediaproxy
by golia vincenzo 10 Apr '06

10 Apr '06

Hi all, i installed ser preconfigured for presence.Now i install mediaproxy for Nat traversal and i have some problems whit ser.cfg Someone can help me or show me an example of ser.cfg? Sorry for my English. Thanks Bye

2 1

[Serusers] help needed
by fay butt 10 Apr '06

10 Apr '06

Hi, I am new user in VOIP, I installed SER with asterisk on two different servers. I installed ser-0.8.12-0.i386.rpm and ser-mysql-0.8.12-0.i386.rpm according to instruction on fedora core 4. I configured the UA's and those are making calls between each other but they are not authenticated by database(mysql). I want to confirm that is there need to seperately install more packages like (authntication,acc,radius) with ser-0.8.12-0.i386.rpm on the ser machine or all the packages are included in the ser-0.8.12-0.i386.rpm file during installation and i just have to configure the ser.cfg. what is the best method to access the database(mysql) with user agents. Best Regards M.Faysal --------------------------------- Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries for just 2¢/min with Yahoo! Messenger with Voice.

1 0

[Users] Radius_extra in ACC
by Deepak Singhal 10 Apr '06

10 Apr '06

Hi All, I want to add the SOURCE IP Address attribute in radius requests of openser. Can someone guide me through the steps needed to achieve this. I have examples for doing the same for DB_extra but unable to achieve the same for radius_extra. Can someone post the cfg lines for achieving this? Deepak Singhal

3 4

[Users] regular expression
by Jayesh Nambiar 10 Apr '06

10 Apr '06

hi all, I want to match the ruri "*86". Can someone please help me with the regular expression to match this. When I do "^sip:*86[0-9]*@", it does not match this expression. Can you also point me to some link which explains well about how to write regular expressions. thank you.. jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

3 2

[Users] lcr SQL query in load_gws
by Nicolas Olivier 10 Apr '06

10 Apr '06

Hi, I recently setup routing on my gateways with the lcr module. I encounter a small problem, quoting from the README: "Matching gateways are then ordered for forwarding purpose (1) according to longest user part match, (2) according to gateway's priority, and (3) randomly." But in fact, and according to the code, the ordering is done firstly on the smallest user part match. Here is a patch, if what I'm say is correct: --- openser-1.0.1-tls/modules/lcr/lcr_mod.c 2005-10-28 12:30:13.000000000 +0200 +++ openser-1.0.1-tls-new/modules/lcr/lcr_mod.c 2006-04-07 14:41:35.000000000 +0200 @@ -715,7 +715,7 @@ from_uri = get_from(_m)->uri; } - q_len = snprintf(query, MAX_QUERY_SIZE, "SELECT %.*s.%.*s, %.*s.%.*s, %.*s.%.*s, %.*s.%.*s, %.*s.%.*s FROM %.*s, %.*s WHERE '%.*s' LIKE %.*s.%.*s AND '%.*s' LIKE CONCAT(%.*s.%.*s, '%%') AND %.*s.%.*s = %.*s.%.*s ORDER BY CHAR_LENGTH(%.*s.%.*s), %.*s.%.*s DESC, RAND()", + q_len = snprintf(query, MAX_QUERY_SIZE, "SELECT %.*s.%.*s, %.*s.%.*s, %.*s.%.*s, %.*s.%.*s, %.*s.%.*s FROM %.*s, %.*s WHERE '%.*s' LIKE %.*s.%.*s AND '%.*s' LIKE CONCAT(%.*s.%.*s, '%%') AND %.*s.%.*s = %.*s.%.*s ORDER BY CHAR_LENGTH(%.*s.%.*s) DESC, %.*s.%.*s DESC, RAND()", gw_table.len, gw_table.s, ip_addr_col.len, ip_addr_col.s, gw_table.len, gw_table.s, port_col.len, port_col.s, gw_table.len, gw_table.s, uri_scheme_col.len, uri_scheme_col.s, Regards, Nicolas

2 2

[Users] no fifo_db_url given - fifo DB commands disabled
by D'Addelfio Davide 10 Apr '06

10 Apr '06

Hello, this is what I have in my syslog file when I running openser, I'd like to know what's mean the last line, what's fifo_db_url and why it is disabled? Apr 10 01:02:30 localhost /usr/sbin/openser[1409]: INFO: fifo process starting: 1409 Apr 10 01:02:30 localhost /usr/sbin/openser[1409]: open_uac_fifo: fifo server up at /tmp/openser_fifo... Apr 10 01:02:30 localhost /usr/sbin/openser[1409]: INFO: no fifo_db_url given - fifo DB commands disabled! Thanks for reply. Davide

2 1

[Users] Prasing SDP messages
by D'Addelfio Davide 10 Apr '06

10 Apr '06

Hello, i need to work over SDP messages extracting from the SIP requests that arrive in my openser. Can it do something like parsing into database of these SDP messages? Thanks in advance for reply Davide

3 2

RE: [Serusers] variables which can be used when writing log
by İlker Aktuna (Koç.net) 10 Apr '06

10 Apr '06

Hi, I need to learn what variables I can use when writing log, for debugging purposes. Can anyone tell me variables that are avilable in logging ? Especially a variable for method name and sip message direction would be very useful. Thanks, ilker ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, April 07, 2006 3:36 PM To: serusers(a)lists.iptel.org Subject: [Serusers] variables which can be used when writing log How can I write the uri and myself variables to log file ? I know that %tu and %fu are used for to uri and from uri but what other variables can I use when writing log ? I need to see the "myself" variable in the log because a uri==myself statement does not match where it should. Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I need to learn what variables I can use when writing log, for debugging purposes. Can anyone tell me variables that are avilable in logging ? Especially a variable for method name and sip message direction would be very useful. Thanks, ilker ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, April 07, 2006 3:36 PM To: serusers(a)lists.iptel.org Subject: [Serusers] variables which can be used when writing log How can I write the uri and myself variables to log file ? I know that %tu and %fu are used for to uri and from uri but what other variables can I use when writing log ? I need to see the "myself" variable in the log because a uri==myself statement does not match where it should. Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/060D0602/04044F00/07064D05/155…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

3 2

[Users] mediaproxy problem
by unplug 10 Apr '06

10 Apr '06

I am using openser with mediaproxy enabled. All device behind NAT can make/receive call without problem. However, when an end device connect to the openser directly using public IP (without NAT), the following problem will occur. -when make outgoing calls, I can hear the receiver but receiver can't hear me -when incoming call, it works as normal Does modiaproxy can handle device using NAT and without NAT? How can I solve the above problem? Is it the port setting problem? Below is the dial plan and the ini file of mediaproxy. http://meerkat.no-ip.com/openser/openser.txt http://meerkat.no-ip.com/openser/mediaproxy.txt Thanks, unplug

1 1

RE: [Serusers] Listen on multiple ports and for multiple domains
by İlker Aktuna (Koç.net) 10 Apr '06

10 Apr '06

Hi, Can you give me an example config for listening on 2 ports and for listening on 2 domains ? Thanks, ilker ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, April 07, 2006 3:08 PM To: Klaus Darilion Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Listen on multiple ports and for multiple domains Hi, In my config file there is no listen directives; only one port= directive. How can I add multiple ports to listen on ? Can you give an example use ? Thanks, ilker -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, April 07, 2006 2:49 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Listen on multiple ports and for multiple domains İlker Aktuna (Koç.net) wrote: > > > Hi, > > Is it possible to force Ser to listen on multiple ports ? > I just tried adding multiple port lines to the config file, but it did > not start that way. > multiple listen=... directives should work. > Also, another question is how to add multiple domains. Again, I tried > to add multiple domain lines to the config but didn't work. you can either use multiple alias= directives or use the domain module and the functions provided by this module. You also have to enable domain support in several modules: modparam("group|usrloc", "use_domain", 1) regards klaus C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Can you give me an example config for listening on 2 ports and for listening on 2 domains ? Thanks, ilker ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, April 07, 2006 3:08 PM To: Klaus Darilion Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Listen on multiple ports and for multiple domains Hi, In my config file there is no listen directives; only one port= directive. How can I add multiple ports to listen on ? Can you give an example use ? Thanks, ilker -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, April 07, 2006 2:49 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Listen on multiple ports and for multiple domains İlker Aktuna (Koç.net) wrote: > > > Hi, > > Is it possible to force Ser to listen on multiple ports ? > I just tried adding multiple port lines to the config file, but it did > not start that way. > multiple listen=... directives should work. > Also, another question is how to add multiple domains. Again, I tried > to add multiple domain lines to the config but didn't work. you can either use multiple alias= directives or use the domain module and the functions provided by this module. You also have to enable domain support in several modules: modparam("group|usrloc", "use_domain", 1) regards klaus <http://387555.sigclick.mailinfo.com/sigclick/03090606/04064E03/02034D01/201…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] Over-logging of the Mediaproxy [wjp]
by Wilko van der Ploeg 10 Apr '06

10 Apr '06

I've installed the mediaserver/ mediaproxy, but I noticed it creates a lot of completely non-informative (only the word 'mediaproxy' ) log entries... some megabytes per test-call... how can I stop this logging ? Wilko van der Ploeg

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Re: [Serusers] pa module fails
by Vaclav Kubart 10 Apr '06

10 Apr '06

I made a snapshot silently (it is on the ftp://ftp.iptel.org/pub/ser/presence/) because I'm going to do more changes there this week (configuration will differ and people always try their old config file and complain if it doesn't work). There are some new things: - authorization of MESSAGEs (not standardized), - storing watchers when presentity is offline and sending them in watcherinfo after presentity subscribes to it - separated XCAP queries from handle_rls_susbcription function - added function for test if presentity is online - RLS module is able to subscribe to external users - more stable PA and RLS modules and a bit optimized - updated presence handbook Will be in next snapshot: - separated XCAP queries into standalone xcap module (libraries and PA and RLS modules will be independent on libcurl) ... done on CVS yet - PA module will be able to subscribe to users (for clients unable to send PUBLISH) Vaclav On Fri, Apr 07, 2006 at 09:53:23PM -0300, Guty wrote: > > Hi, do you have any news about new presence snapshot? > Is there anything new? > Saludos, > Gustavo. > > > -----Mensaje original----- > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Enviado el: martes, 21 de marzo de 2006 12:08 > Para: Guty > CC: serusers(a)lists.iptel.org > Asunto: Re: [Serusers] pa module fails > > I think, that presence modules are only in source version - there are > not debian packages with them (the best is CVS head). Please wait > shortly for new presence snapshot... > Vaclav > > On Tue, Mar 21, 2006 at 11:52:34AM -0300, Guty wrote: > > I'm using ser 0.9.6. First I try with the stable version of all packages, > > didn't work, then I downloaded different versions of pa module, unstable, > > testing, sarge... but the error is still the same. > > Do you know any configuration that work with presence? > > > > > > -----Mensaje original----- > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > > Enviado el: martes, 21 de marzo de 2006 3:37 > > Para: Guty > > CC: serusers(a)lists.iptel.org > > Asunto: Re: [Serusers] pa module fails > > > > You probably use old ser version. Don't you? I think, that presence > > modules in stable ser versions were always old. > > Vaclav > > > > On Mon, Mar 20, 2006 at 12:23:30PM -0300, Guty wrote: > > > Hi, > > > > > > I've configured my Ser for presence handling, but i get some error son > my > > > ser.cfg file: > > > > > > > > > > > > 0(1761) set_mod_param_regex: pa matches module pa > > > > > > 0(1761) set_mod_param_regex: parameter <auth> not found in module <pa> > > > > > > 0(1761) parse error (75,20-21): Can't set module parameter > > > > > > 0(1761) set_mod_param_regex: pa matches module pa > > > > > > 0(1761) set_mod_param_regex: parameter <winfo_auth> not found in module > > > <pa> > > > > > > 0(1761) parse error (76,20-21): Can't set module parameter > > > > > > > > > > > > My file includes the following: > > > > > > > > > > > > loadmodule "/usr/lib/ser/modules/pa.so" > > > > > > > > > > > > > > > ---------------------------------------------------------------------------- > > > ---------------------- > > > > > > > > > > > > # -- pa params -- > > > > > > modparam("pa", "use_db", 1) > > > > > > modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") > > > > > > modparam("pa", "auth", "none") > > > > > > modparam("pa", "winfo_auth", "none") > > > > > > > > > > > > > > > ---------------------------------------------------------------------------- > > > ---------------------- > > > > > > > > > > > > if (method=="SUBSCRIBE") { > > > > > > if (!t_newtran()) { > > > > > > sl_reply_error(); > > > > > > }; > > > > > > log(1, "Handle Subscribe"); > > > > > > handle_subscription("registrar"); # uses uid from AVP > > > > > > (get_to_uid) > > > > > > break; > > > > > > }; > > > > > > > > > > > > if (method=="PUBLISH") { > > > > > > if (!t_newtran()) { > > > > > > log(1, "newtran error\n"); > > > > > > sl_reply_error(); > > > > > > }; > > > > > > log(1, "Handle Publish"); > > > > > > handle_publish("registrar"); > > > > > > break; > > > > > > }; > > > > > > I've installed SER 0.9.6-0.1 for Debian , ser-mysql-module 0.9.6-0.1 for > > > Debian, and ser-pa-module 0.9.6-0.1 for Debian > > > > > > Any idea of what is going wrong? > > > > > > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] OpenSER Training
by Dioris Moreno 10 Apr '06

10 Apr '06

Does anybody know about existing OpenSER training? Thanks

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RE: [Serusers] error when running SERCTL
by Gould, Aaron 08 Apr '06

08 Apr '06

is it ok if i put "setenv SIP_DOMAIN lab.com" in my csh .cshrc file ? this is what i did and it seems to have fixed it. i can run serctl now. is that ok to do it that way? aaron ________________________________ From: PREVOT Paul [mailto:prevot@enseirb.fr] Sent: Fri 4/7/2006 11:36 AM To: Gould, Aaron; serusers(a)lists.iptel.org Subject: RE: [Serusers] error when running SERCTL Export SIP_DOMAIN="yourdomain" or "xxx.xxx.xxx.xxx" -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Gould, Aaron Envoyé : vendredi 7 avril 2006 19:34 À : serusers(a)lists.iptel.org Objet : [Serusers] error when running SERCTL i get an error when i try to run SERCTL. does anyone know how to fix it? the message says.... "You need to set enviorment variable SIP_DOMAIN (e.g. to 'foobar.com') first." aaron _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Limit number of calls
by zakforever＠netscape.net 08 Apr '06

08 Apr '06

Hi All, Is there a way in SER that would limit the number of calls that any user can make at any one time to just one call ? Zak ___________________________________________________ Try the New Netscape Mail Today! Virtually Spam-Free | More Storage | Import Your Contact List http://mail.netscape.com

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[Serusers] error when running SERCTL
by Gould, Aaron 07 Apr '06

07 Apr '06

i get an error when i try to run SERCTL. does anyone know how to fix it? the message says.... "You need to set enviorment variable SIP_DOMAIN (e.g. to 'foobar.com') first." aaron

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[Users] method INVITE and module ACC
by buxalex＠libero.it 07 Apr '06

07 Apr '06

Hello, me again.. :o) I set up the openser.cfg as follow: loadmodule "/usr/local/lib/openser/modules/acc.so" #----------------------------------- # ACC parameters #----------------------------------- modparam("acc", "db_url", "mysql://openser:openser@localhost/openser") modparam("acc", "failed_transaction_flag", 4) modparam("acc", "report_cancels", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_level", 1) modparam("acc", "log_fmt", "miocfs") ..... if (method=="REGISTER") { if (!www_authorize("161.27.53.93", "subscriber")) { www_challenge("161.27.53.93", "0"); exit; }; save("location"); exit; }; if (method=="INVITE") { setflag(1); record_route(); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } my question is: It is enough to have the ACC module loaded and configured, the setup of the INVITE method as above, to have the db updated? Or I completely misunderstood the manual? This because with this configuration I can perform a call between two softphones, but I dont have any table filled into the database. Thanks a lot Alessandro

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[Serusers] How can i setup SER on Cygwin?
by Garima Mishra 07 Apr '06

07 Apr '06

i am trying to setup SER on cygwin but during compilation i am getting errors like "Makefile.rules:80: qvalue.d: No such file or directory" same error for many files... can anyone suggest me how can i setup SER on cygwin? Thanks, Garima Garima Mishra Software Engineer <http://www.openeratech.com/> <http://www.openeratech.com/> <http://www.openeratech.com/> Openera Technologies Pvt Ltd. 97, Block 5 Koramangala Indl. Estate <?xml:namespace prefix = st1 ns = "urn:schemas-microsoft-com:office:smarttags" />Bangalore - 560 095. India Tel: +91 80 2552 1937 Fax: +91 80 2552 1164 <http://www.openeratech.com/> www.openeratech.com E-mail: <mailto:garima@openeratech.com> garima(a)openeratech.com This message (including any attachments) is a confidential and Privileged communication to the intended addressee. If you are not the Intended addressee, you have received this message in error. In that case, please permanently delete this message and notify the sender so that we can avoid this inconvenience in the future. Thank you.

2 1

Re: [Users] aut_db module and realm
by buxalex＠libero.it 07 Apr '06

07 Apr '06

hello Bogdan, I made a mistake into openser.cfg. I didnt set the use_domain for the registrar module. Now everything is fine and the domain is stored into location table. thanks a lot to evrybody for your support. Alex ---------- Initial Header ----------- >From : "Bogdan-Andrei Iancu" bogdan(a)voice-system.ro To : "buxalex(a)libero.it" buxalex(a)libero.it Cc : "christophe.irles" christophe.irles(a)alcatel.fr,"users" users(a)openser.org Date : Fri, 07 Apr 2006 17:17:30 +0300 Subject : Re: [Users] aut_db module and realm > Hi, > > what do you mean by crash - cannot start due some errors (see the logs) ? > > regards, > bogdan > > buxalex(a)libero.it wrote: > > >Hello, > > > >thanks for your help. I entered a new line in my openser.cfg: > > > >modparam("usrloc", "use_domain", 1) > > > >but when I start openser (through Monit service manager) it crashes. > >If I set the parameter value=0 it works fine but, of course, the domain is not populated into location table. > > > >Could you please give me any advice? > > > >Thanks > >Alessandro > >---------- Initial Header ----------- > > > >From : "Bogdan-Andrei Iancu" bogdan(a)voice-system.ro > >To : "Christophe Irles" christophe.irles(a)alcatel.fr > >Cc : buxalex(a)libero.it,"users" users(a)openser.org > >Date : Thu, 06 Apr 2006 19:34:51 +0300 > >Subject : Re: [Users] aut_db module and realm > > > > > > > > > > > > > > > > > > > >>Hi, > >> > >>just as an add-on...to have the domain column populated in location > >>table, you need to set the "use_domain" param in location module. See: > >> http://openser.org/docs/modules/1.0.x/usrloc.html#AEN195 > >> > >>regards, > >>bogdan > >> > >>Christophe Irles wrote: > >> > >> > >> > >>>Hello, > >>> > >>>command [ modparam("auth_db", "domain_column", "domain") ] is used to > >>>indicate to auth_db module the name of the column to use to retrieve the > >>>domain associated to an user in the "subcriber" table (table name use > >>>generally by default in auth_db). > >>> > >>>This table is filled by the administrator to grant access > >>> > >>>The "location" is filled by openSER when user registered on openSER based of > >>>info given by UA. So it's possible to have location.domain != > >>>subscriber.domain if the settings of the UA is not performed correctly. > >>> > >>>Field location.received contains this data SIP:IP:PORT of the UA device > >>> > >>>Hope this helps, > >>>Christophe > >>> > >>>-----Message d'origine----- > >>>De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part > >>>de buxalex(a)libero.it > >>>Envoyé : jeudi 6 avril 2006 16:38 > >>>À : users > >>>Objet : [Users] aut_db module and realm > >>> > >>>Hello to everybody, > >>> > >>>I configured the openser.cfg as follow: > >>> > >>>#----------------------------------- > >>># AUT_DB parameters > >>>#---------------------------------- > >>> > >>>modparam("auth_db", "db_url", "mysql://openser:openser@localhost/openser") > >>>modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", > >>>"domain_column", "domain") modparam("auth_db", "user_column", "phone") > >>>modparam("auth_db", "password_column", "password") > >>> > >>>In this way openser checks for authentication the phone number instead of > >>>the username (it works fine!!). My problem is realted to domain. Entering > >>>the modparam(.....,"domain") I was expecting to see the domain column > >>>populated in the table location, but it doesn't happen. Do I have to > >>>configure something else? Moreover, anybody could explain me the meaning of > >>>the column location.received? > >>> > >>>thanks in advance > >>>Alessandro > >>> > >>> > >>>_______________________________________________ > >>>Users mailing list > >>>Users(a)openser.org > >>>http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >>>_______________________________________________ > >>>Users mailing list > >>>Users(a)openser.org > >>>http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >>> > >>> > >>> > > > > > > > > > >

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Re: [Users] aut_db module and realm
by buxalex＠libero.it 07 Apr '06

07 Apr '06

Hello, thanks for your help. I entered a new line in my openser.cfg: modparam("usrloc", "use_domain", 1) but when I start openser (through Monit service manager) it crashes. If I set the parameter value=0 it works fine but, of course, the domain is not populated into location table. Could you please give me any advice? Thanks Alessandro ---------- Initial Header ----------- >From : "Bogdan-Andrei Iancu" bogdan(a)voice-system.ro To : "Christophe Irles" christophe.irles(a)alcatel.fr Cc : buxalex(a)libero.it,"users" users(a)openser.org Date : Thu, 06 Apr 2006 19:34:51 +0300 Subject : Re: [Users] aut_db module and realm > Hi, > > just as an add-on...to have the domain column populated in location > table, you need to set the "use_domain" param in location module. See: > http://openser.org/docs/modules/1.0.x/usrloc.html#AEN195 > > regards, > bogdan > > Christophe Irles wrote: > > >Hello, > > > >command [ modparam("auth_db", "domain_column", "domain") ] is used to > >indicate to auth_db module the name of the column to use to retrieve the > >domain associated to an user in the "subcriber" table (table name use > >generally by default in auth_db). > > > >This table is filled by the administrator to grant access > > > >The "location" is filled by openSER when user registered on openSER based of > >info given by UA. So it's possible to have location.domain != > >subscriber.domain if the settings of the UA is not performed correctly. > > > >Field location.received contains this data SIP:IP:PORT of the UA device > > > >Hope this helps, > >Christophe > > > >-----Message d'origine----- > >De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part > >de buxalex(a)libero.it > >Envoyé : jeudi 6 avril 2006 16:38 > >À : users > >Objet : [Users] aut_db module and realm > > > >Hello to everybody, > > > >I configured the openser.cfg as follow: > > > >#----------------------------------- > ># AUT_DB parameters > >#---------------------------------- > > > >modparam("auth_db", "db_url", "mysql://openser:openser@localhost/openser") > >modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", > >"domain_column", "domain") modparam("auth_db", "user_column", "phone") > >modparam("auth_db", "password_column", "password") > > > >In this way openser checks for authentication the phone number instead of > >the username (it works fine!!). My problem is realted to domain. Entering > >the modparam(.....,"domain") I was expecting to see the domain column > >populated in the table location, but it doesn't happen. Do I have to > >configure something else? Moreover, anybody could explain me the meaning of > >the column location.received? > > > >thanks in advance > >Alessandro > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

2 1

[Serusers] help with SER on Solaris 10 and softphone on Win2000
by Gould, Aaron 07 Apr '06

07 Apr '06

i have a softphone on a windows 2000 laptop after the softphone sends a "register" message to SER, i see on the sniffer that SER sends a "register" message right back to the laptop. is this normal? and i see next that the laptop sends a SIP 405 "Method Not Allowed" message and that the SER sends the same thing back to the laptop what does this mean? aaron

2 1

[Serusers] strip realm_prefix & radius
by Kostas Marneris 07 Apr '06

07 Apr '06

Hi, I use RADIUS for user authentication. The SER configuration in REGISTER handler: if (!radius_www_authorize("")) { log(1, " - Radius Auth failed - Send challenge to UA"); www_challenge("", "0"); break; }; UAs are configured to use the realm 'domain.com'. There are cases (many UAs does not support DNS SRV) where UAs send 'sip.domain.com'. Which is the way to strip 'sip.' from realm and send it to RADIUS. I used the following : modparam("auth", "realm_prefix", "sip.") but it doesn't work. Actually, SER sends back '401 Unauthorized' with digest.realm="domain.com", and then UA sends REGISTER with realm="domain.com" but 'radius_www_authorize' fail. I see no RADIUS request to reach my RADIUS server. Any idea / help ? thank you Kosats

1 0

[Serusers] What Media Gateway (connected via SS7) do you use
by carlos ivan grajales 07 Apr '06

07 Apr '06

Hi, Cisco AS5300 is a good option. Carlos Grajales __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] logon problem
by unplug 07 Apr '06

07 Apr '06

I have set an openser in a cs department with almost 50 sip phones connected to it. It is a very heavy loading environment. Every sip phone may make/receive almost 20-40 calls per minute. In such situation, most of the sip phones will experience a logon problem. They will failed to logon (in the display of the sip phones show wait logon) and 1 minute after, they can logon to the system. This situation will repeat for every sip phone in the system. I wonder what is capacity of the openser. As I expect, it should handle over 1000 sip devices connected concurrently (correct me if I am wrong). I also wonder any misconfigure in the configuration file. I set children to 4 in the file. Do I need to increase the children value? Is it useful for solve this situation? If not, what is the possible action to solve the above problem?

4 3

[Serusers] variables which can be used when writing log
by İlker Aktuna (Koç.net) 07 Apr '06

07 Apr '06

How can I write the uri and myself variables to log file ? I know that %tu and %fu are used for to uri and from uri but what other variables can I use when writing log ? I need to see the "myself" variable in the log because a uri==myself statement does not match where it should. Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/000E0500/04044803/02034E07/236…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

RE: [Serusers] Listen on multiple ports and for multiple domains
by İlker Aktuna (Koç.net) 07 Apr '06

07 Apr '06

Hi, In my config file there is no listen directives; only one port= directive. How can I add multiple ports to listen on ? Can you give an example use ? Thanks, ilker -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, April 07, 2006 2:49 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Listen on multiple ports and for multiple domains İlker Aktuna (Koç.net) wrote: > > > Hi, > > Is it possible to force Ser to listen on multiple ports ? > I just tried adding multiple port lines to the config file, but it did > not start that way. > multiple listen=... directives should work. > Also, another question is how to add multiple domains. Again, I tried > to add multiple domain lines to the config but didn't work. you can either use multiple alias= directives or use the domain module and the functions provided by this module. You also have to enable domain support in several modules: modparam("group|usrloc", "use_domain", 1) regards klaus <http://387555.sigclick.mailinfo.com/sigclick/03090505/04064D03/01054E02/201…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Serusers] Listen on multiple ports and for multiple domains
by İlker Aktuna (Koç.net) 07 Apr '06

07 Apr '06

Hi, Is it possible to force Ser to listen on multiple ports ? I just tried adding multiple port lines to the config file, but it did not start that way. Also, another question is how to add multiple domains. Again, I tried to add multiple domain lines to the config but didn't work. Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/060A0407/01044E00/04004F00/152…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] MediaProxy with 2 IPs
by Joao Pereira 07 Apr '06

07 Apr '06

Hello to all Im using RTPproxy, with 2 IP addresses (one private and one public IP), this way: "rtpproxy -l private_IP/public_IP" The calls from the LAN to the Internet (and vice versa) must pass through the server, (from the private IP to the public IP) . This is made with NatHelper + RTPproxy. But Im having some problems with RTPproxy and I would like to try MediaProxy. Can MediaProxy work with 2 IPs (one private and one public)? Thanks Joao Pereira

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[Serusers] (no subject)
by Jan Bollen 07 Apr '06

07 Apr '06

Hi, Last week when I installed SER on Debian, I encounterd the same problem. The MySQl manual gives you a couple of courses of actions: See http://downloads.mysql.com/docs/refman-4.1-en.a4.pdf section A.2.3 ------------------------------------------------------------ A.2.3. Client does not support authentication protocol MySQL 4.1 and up uses an authentication protocol based on a password hashing algorithm that is incompatible with that used by older clients. If you upgrade the server to 4.1, attempts to connect to it with an older client may fail with the following message: shell> mysql Client does not support authentication protocol requested by server; consider upgrading MySQL client To solve this problem, you should use one of the following approaches: 1 Upgrade all client programs to use a 4.1.1 or newer client library. 2 When connecting to the server with a pre-4.1 client program, use an account that still has a pre-4.1-style password. **3 Reset the password to pre-4.1 style for each user that needs to use a pre-4.1 client program. This can be done using the SET PASSWORD statement and the OLD_PASSWORD() function: mysql> SET PASSWORD FOR -> 'some_user'@'some_host' = OLD_PASSWORD('newpwd'); Substitute the password you want to use for �newpwd� in the preceding examples. MySQL cannot tell you what the original password was, so you'll need to pick a new one. - Tell the server to use the older password hashing algorithm: ** 1. Start mysqld with the --old-passwords option. ------------------------------------------------------------ The actions marked with ** above is what I did and after restarting MySQl it worked fine. cheers, Jan =========================================================== Hi again, Thank you for your quick answer. Here is my error log output. Serweb [error] file:/usr/home/dev/serweb-0.9.4/html/data_layer.php:465 DB Error: connect failed - [native code = Client does not support authentication protocol requested by server; consider upgrading MySQL client] ** mysql://ser:heslo@localhost/ser When i read MySQL Reference Manual i see that, MySQL changed its password style in 4.1 or newer versions. (in Serweb install manual it writes MySQL 4.0, i didn't think that it could be a problem and installed 4.1.13) What do you advice me to do? Gokay

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[Users] Which RFC openser support ?
by Nguyen Duc Phi 07 Apr '06

07 Apr '06

Hello, Could you let me know "Does Operser support the following RFC?" RFC 2445 Time definition RFC 2869 RADIUS Extensions RFC 2868 RADIUS Attributes for Tunnel Protocol Support RFC 2867 RADIUS Accounting Modifications for Tunnel Protocol Support Best regards, Nguyen

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Re: Re: [Serusers] Serweb problem
by Gokay Tosunoglu 07 Apr '06

07 Apr '06

Hi again, Thank you for your quick answer. Here is my error log output. Serweb [error] file:/usr/home/dev/serweb-0.9.4/html/data_layer.php:465 DB Error: connect failed - [native code = Client does not support authentication protocol requested by server; consider upgrading MySQL client] ** mysql://ser:heslo@localhost/ser When i read MySQL Reference Manual i see that, MySQL changed its password style in 4.1 or newer versions. (in Serweb install manual it writes MySQL 4.0, i didn't think that it could be a problem and installed 4.1.13) What do you advice me to do? Gokay Hi, the problem is, that serweb cannot connect to your mysql. Check your serweb log (logging have to be enabled in serweb config). There should be more descriptive info. Karel Gokay Tosunoglu napsal(a): > Hi, > I'm a new ser user and have a problem with serweb. > I installed a new system of Freebsd 6.0, apache 1.3.33, php 4.3.11, > mysql 4.1.13, ser-0.9.6 and serweb-0.9.4 . > All configurations are seem to be ok. (ser_mysql.sh is ok, too.) I enter > the admin login screen of serweb and write admin:heslo as > username:password. The js automatically completing the username to my > default domain so it becomes "admin(a)multitekarge.com" and i can't login. > The error message is : > " > DB Error: connect failed > Bad username or password > " > Altough for database, name is "ser", user is "ser" and pass is "heslo" > in config_data_layer.php, it cannot connect. > > I also added a new user to ser by typing > " > serctl add gokay(a)multitekarge.com gokay gokay.tosunoglu(a)avrupayazilim.com > " > and tried to enter with that user in user login screen. I got the same > error. > > I'll got lot of pleasure of any help. > Thanks, > Gokay > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Problem with forwarding - 480 response.
by Tomasz Stafin 07 Apr '06

07 Apr '06

Helo Thanks for help but my gateway is isdn gateway. This kind of gateway send a response "486" to server when user is busy and 480 when isdn is offline. T_on_failure will forward a transaction when 486 return but i wanted to forward only if 480 response return to server. Thanks for help. TS

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[Serusers] Ser as a presence server with other SIP prox as registrar and rtp proxy
by İlker Aktuna (Koç.net) 07 Apr '06

07 Apr '06

Hi, I am using the following routing configuration for testing Ser as a presence server only. The aim is to route all register, invite and other audio related messages to the main SIP and RTP proxy while keeping presence messages to SER. But there are some configuration parts that I don't understand: 1- if I remove the following lines from route[2] , client can't register through Ser. if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; 2- What is the meaning of : if (uri==myself) { ? I just commented this out and it seems to work. 3- Can anyone explain me the meaning of following lines: # if (!method=="REGISTER") record_route(); # if (loose_route()) { # # mark routing logic in request # append_hf("P-hint: rr-enforced\r\n"); # route(1); # break; # }; Thanks, İlker Below is my full routing configuration: # main routing logic route{ # XML RPC if (method == "POST" || method == "GET") { create_via(); dispatch_rpc(); break; } # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing # if (loose_route()) { # # mark routing logic in request # append_hf("P-hint: rr-enforced\r\n"); # route(1); # break; # }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # if (uri==myself) { # if (!lookup_domain("To")) { #xlog("L_ERR", "URI=MYSELF\n"); #xlog("L_ERR", "to: %tu from: %fu\n"); # log(1, "Unknown domain - message should be forwarded?"); # route(1); # break; # } # } if (method=="REGISTER") { xlog("L_ERR", "REGISTER\n"); xlog("L_ERR", "to: %tu from: %fu\n"); append_hf("P-hint: outbound\r\n"); save_noreply("location"); route(2); break; }; if (method=="INVITE") { xlog("L_ERR", "INVITE\n"); xlog("L_ERR", "to: %tu from: %fu\n"); if (!lookup("location")) { route(2); break; }; if (lookup("location")) { route(1); break; }; }; if (method=="MESSAGE") { xlog("L_ERR", "MESSAGE\n"); xlog("L_ERR", "to: %tu from: %fu\n"); if (!lookup("location")) { sl_send_reply("404","Not found"); break; }; route(1); break; }; if (method=="NOTIFY") { xlog("L_ERR", "NOTIFY\n"); xlog("L_ERR", "to: %tu from: %fu\n"); if (!lookup("location")) { sl_send_reply("404","Not found"); break; }; route(1); break; }; if (method=="SUBSCRIBE") { xlog("L_ERR", "SUBSCRIBE\n"); xlog("L_ERR", "to: %tu from: %fu\n"); if (!t_newtran()) { # creates a new transaction sl_reply_error(); break; }; handle_subscription("registrar"); break; }; if (method=="PUBLISH") { xlog("L_ERR", "PUBLISH\n"); xlog("L_ERR", "to: %tu from: %fu\n"); if (!t_newtran()) { sl_reply_error(); break; }; handle_publish("registrar"); break; }; xlog("L_ERR", "OTHER\n"); xlog("L_ERR", "to: %tu from: %fu\n"); if (!lookup("location")) { route(2); break; }; if (lookup("location")) { route(1); break; }; # append_hf("P-hint: outbound\r\n"); # route(2); # break; } route[1] { xlog("L_INFO","t_relay()\n"); if (!t_relay()) { sl_reply_error(); }; } route[2] { if (!t_newtran()) { xlog("L_ERR", "newtran error ?????\n"); log(1, "newtran error\n"); sl_reply_error(); break; }; if (!t_forward_nonack("10.100.200.97", 5060)) { sl_reply_error(); }; xlog("L_ERR", "FORWARD\n"); } <http://387555.sigclick.mailinfo.com/sigclick/04090107/04074E00/07094A00/171…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] Need urgent help/pointers reg. Mediaproxy
by Dave 07 Apr '06

07 Apr '06

I had posted the following a few days ago and received no response. Can someone please help! Hello,I have the following setup: UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW | IVR Nat'd user calls a number that is forwarded (t_relay) to Prepaid and the user is prompted for PIN#/Destination Phone# and all works fine. Mediaproxy is invoked and a mediaproxy port# (35774) is assigned during the whole process. Once the user enters the destination number, the Prepaid forwards the call to the PSTNGW and the PSTN phone rings (Call signalling working fine). When the PSTN phone is answered, no media can be heard!!! I did some investigation and here is what is happening: Once the user enters the destination# to dial, The Prepaid/B2BUA does two things: 1) Sends a re-invite to put the UA on hold. 2) Sends an INVITE to the PSTN GW and retrieves the SDP from the response from PSTN GW 3) Sends a second-reinvite to the UA via SER with the SDP info of the PSTNGW 4) SER invokes the mediaproxy (since it is reinvite) and assigns the SAME MEDIAPROXY PORT# as earlier on when the media was flowing fine (SDP has audio port# 35774) 5) SER forwards the re-INVITE TO UA 6) UA responds with a 200 OK and sends 200 OK to SER. SER agains invokes mediaproxy and assigns the SAME MEDIAPROXY PORT# (audio port 35774) and forwards the response to Prepaid (which sends it to PSTNGW). I checked on Cisco PSTNGW that it is creating a session with the mediaproxy (port:35774). BUT NO AUDIO CAN BE HEARD IN ANY DIRECTION. Can anyone please help? Is this a mediaproxy issue that when reinvites are sent and mediaproxy is invoked multiple times, issues arise? I am running the latest mediaproxy version. --------------------------------- Have a question? Yahoo! Canada Answers. Go to Yahoo! Canada Answers

2 3

[Users] comparing an avp to a null value
by Tavis P 07 Apr '06

07 Apr '06

I've been trying to make a comparison, using avp_check, between a pseudo variable (copied to an avp) and a null value (which is a valid state for some pseudo varibles currently) I havn't had any luck in being able to determine if an avp is null short of using a regex (which i try to avoid) so i thought i would mention this here, perhaps this functionality may be seen as useful to others or simply useful in general (tm) Here is a quick example of what i'm currently doing ---- # Store the destination IP into an AVP avp_printf("i:450", "$dd"); In this case, $dd is null because the destination set contains a domain for this branch avp_printf("i:451", "$si"); if ( avp_check("i:450", "re/^.{0,0}$/g") ) { xlog("L_WARN", "[$fU] Route{4}: AVP I:450 was empty"); } ---- I think it was probably thought that an avp should never have a null value therefor there was no logic added to any of the functions used to check/print/write avps which allowed for the creation/evaluation of NULL, however it would be advantageous in the least to include this functionality somewhere (other than regex matches) a few things that i've tried, that were no successful ---- # Store the destination IP into an AVP avp_printf("i:450", "$dd"); avp_printf("i:451", "$si"); avp_printf("i:452", ""); This will break script compilation avp_write("i:452", ""); This will break script compilation avp_write("i:452", "s:"); This will break script compilation if ( avp_check("i:450", "eq//g") ) { xlog("L_WARN", "[$fU] Route{4}: AVP I:450 was empty"); } if ( avp_check("i:450", "eq/0x/g") ) { xlog("L_WARN", "[$fU] Route{4}: AVP I:450 was empty"); } if ( avp_check("i:450", "eq/i:452/g") ) { xlog("L_WARN", "[$fU] Route{4}: AVP I:450 was empty"); } ----

2 1

[Users] Error using subst function
by Dmitry Lyubimkov 07 Apr '06

07 Apr '06

We need to make 2 transformations in URI and To field 1. Convert telephone number to to format E.164 2. To correct a domain part of the address if it is set incorrectly by stupid UA All this is done through set of if operators because every of transformations can be absent or can be done on own algorithm. Let's consider a special case. We have on an input 799111(a)62.33.22.14 and on an output should receive 78142799111(a)voapp.ru URI will be transformed by these commands subst_uri("/^sip:([0-9]+)@(.*)/sip:78142\1@\2/i"); subst_uri("/^sip:([0-9]+)@(.*)/sip:\1@voapp.ru/i"); To field will be transformed by these commands subst("/^To:(.*)sip:([0-9]+)@(.*)/To:\1sip:78142\2@\3/i"); subst("/^To:(.*)sip:([0-9]+)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/ To:\1sip:\2@voapp.ru\7/i"); We test and get dump of messages using tcpdump We receive from UA INVITE sip:705353@62.33.22.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.13.15 From: ......._1 <sip:gan@192.168.13.15>;tag=22683200-29298-10909 To: <sip:705353@62.33.22.14> ............ We make transformation and sent following: INVITE sip:78142705353@voapp.ru SIP/2.0 Max-Forwards: 10 Record-Route: <sip:62.33.22.14;ftag=22683200-29298-10909;lr> Via: SIP/2.0/UDP voapp.ru:5060;branch=z9hG4bKa4db.e5fa9a02.0 Via: SIP/2.0/UDP 192.168.13.15;rport=53809;received=217.107.59.194 From: ......._1 <sip:gan@192.168.13.15>;tag=22683200-29298-10909 To: <sip:78142705353@62.33.22.14> To: <sip:705353@voapp.ru> ............... As you can see URI was transformed correctly And instead of To field we have 2 To fields everyone corresponding to one of transformations. It is a bug? Or it is feature OpenSER? Whether it is possible to bypass this restriction? I understand that it is possible to make one subst call instead of two calls. subst("/^To:(.*)sip:([0-9]+)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/ To:\1sip:78142\2@voapp.ru\7/i"); But in a reality the algorithm is much more complex. And that instead of several consecutive transformations to leave one it is required to use a many of if operators. It is very inconvenient, conducts to mistakes and reduces readability of a code. Dmitry

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[Serusers] Serweb problem
by Gokay Tosunoglu 07 Apr '06

07 Apr '06

Hi, I'm a new ser user and have a problem with serweb. I installed a new system of Freebsd 6.0, apache 1.3.33, php 4.3.11, mysql 4.1.13, ser-0.9.6 and serweb-0.9.4 . All configurations are seem to be ok. (ser_mysql.sh is ok, too.) I enter the admin login screen of serweb and write admin:heslo as username:password. The js automatically completing the username to my default domain so it becomes "admin(a)multitekarge.com" and i can't login. The error message is : " DB Error: connect failed Bad username or password " Altough for database, name is "ser", user is "ser" and pass is "heslo" in config_data_layer.php, it cannot connect. I also added a new user to ser by typing " serctl add gokay(a)multitekarge.com gokay gokay.tosunoglu(a)avrupayazilim.com " and tried to enter with that user in user login screen. I got the same error. I'll got lot of pleasure of any help. Thanks, Gokay

2 1

[Serusers] ser /tmp/webfifo errors, please help me
by mkumar＠mantragroup.com 07 Apr '06

07 Apr '06

Hi All, Everything was working fine until today when I started getting errors like below but still I am able to make and receive calls. 1(9504) ERROR: open_reply_pipe: open error (/tmp/webfifo_452863019): No such file or directory 1(9504) ERROR: fifo_stats: file /tmp/webfifo_452863019 bad: No such file or directory 1(9504) ERROR: fifo_server: command (t_stats) processing failed 1(9504) ERROR: open_reply_pipe: open error (/tmp/webfifo_109612442): No such file or directory 1(9504) ERROR: fifo_stats: file /tmp/webfifo_109612442 bad: No such file or directory 1(9504) ERROR: fifo_server: command (t_stats) processing failed Some settings with my machine might have changed as we worked on some system changes. Please tell me the reason for this errors and how to fix them? Thanks, Manoj.

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[Serusers] Problem with the HA Gateway
by Teran Jesus 06 Apr '06

06 Apr '06

Hello I Have a problem with my gatewas, because I want if one Gateway is down sends to the other gateway i implemented something like this: rewritehost("GATEWAY1"); t_on_failure("1"); t_relay(); And in failure 1: rewritehost("GATEWAY2"); append_branch(); t_relay(); In the INVITE works fine because if GATEWAY1 is down or something is bad, this sends a error and the INVITE is send to the GATEWAY2, but in the ACK is the problem because the ACK allways is send to the GATEWAY1 and this dont respond with an error or something like that. Do you know how can I do to resolv this problem? Regards Atentamente: Ing. Jesús Terán Blásquez MCM Telecomunicaciones 53.50.00.57 Planeación

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[Serusers] Problem with forwarding - 480 response.
by Tomasz Stafin 06 Apr '06

06 Apr '06

Helo everyone! I've a problem with forwarding. My gateway return a "480" response when line is offline and I'm trying to forward to another gateway when line is offline (480 is return). I don't want to use t_on_failure. I write something like this: if (uri="sip:0@.*"){ sethost("192.168.10.1") t_relay(); if (t_check_status ("480")) { sethost("192.168.10.2"); t_relay(); break; } break; } But this script dont't work. I want to forward all calls only if return 480 sip response (not 486). Thanks for any help. Sorry for my bad english. Swaper

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[Serusers] Radius Accounting Information
by Alberto Cruz 06 Apr '06

06 Apr '06

Hi everybody. As we know SER/RadiusClient sent the accounting information in the following way: 04/05/2006 22:25:12 Acct-Status-Type = Start 04/05/2006 22:25:12 User-Service-Type = 15 04/05/2006 22:25:12 Sip-Response-Code = 200 04/05/2006 22:25:12 Sip-Method = 1 04/05/2006 22:25:12 User-Name = "55999999(a)192.168.75.4" 04/05/2006 22:25:12 Calling-Station-Id = "sip:55999999@192.168.75.4" 04/05/2006 22:25:12 Called-Station-Id = "sip:015527350900@192.168.75.4" 04/05/2006 22:25:12 Sip-Translated-Request-URI = "sip:015527350900@201.37.23.105" 04/05/2006 22:25:12 Acct-Session-Id = "3565c759-7f2a08a6(a)172.16.1.50" 04/05/2006 22:25:12 Sip-To-Tag = "1FFFF924-24EF" 04/05/2006 22:25:12 Sip-From-Tag = "4c676fbdf6dbdb92o0" 04/05/2006 22:25:12 Sip-Cseq = "101" 04/05/2006 22:25:12 Client-Port-Id = 5060 04/05/2006 22:25:12 Acct-Delay-Time = 0 04/05/2006 22:25:12 Client-Id = 192.168.75.4 Is there a way at SER accounting parameters or Radius Client Configuration that we can give a format to the Calling-Station-Id/Called-Station-Id in order to strip the "sip:" string? Regards Alberto Cruz

1 0

[Users] Basic and digest authentication
by Christophe Irles 06 Apr '06

06 Apr '06

Hi, As I understood the SIP authentication process is based on RFC2617 i.e. the same authentication process used in HTTP. So both Basic and Digest authentication are possible. Is-it possible to force always digest authentication ? The goal is to avoid to have credential in clear if basic authentication is used. It's possible on web server so I supposed it's possible also in SIP. Perhaps this feature is by default on (open)SER since I never viewed in my tests passwords in clear. Thanks in advance, Christophe

2 1

[Users] aut_db module and realm
by buxalex＠libero.it 06 Apr '06

06 Apr '06

Hello to everybody, I configured the openser.cfg as follow: #----------------------------------- # AUT_DB parameters #---------------------------------- modparam("auth_db", "db_url", "mysql://openser:openser@localhost/openser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "domain_column", "domain") modparam("auth_db", "user_column", "phone") modparam("auth_db", "password_column", "password") In this way openser checks for authentication the phone number instead of the username (it works fine!!). My problem is realted to domain. Entering the modparam(.....,"domain") I was expecting to see the domain column populated in the table location, but it doesn't happen. Do I have to configure something else? Moreover, anybody could explain me the meaning of the column location.received? thanks in advance Alessandro

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[Serusers] Help with MediaProxy/B2bUA (Urgent - Pls. help)
by Dave 06 Apr '06

06 Apr '06

Hello,I have the following setup: UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW | IVR Nat'd user calls a number that is forwarded (t_relay) to Prepaid and the user is prompted for PIN#/Destination Phone# and all works fine. Mediaproxy is invoked and a mediaproxy port# (35774) is assigned during the whole process. Once the user enters the destination number, the Prepaid forwards the call to the PSTNGW and the PSTN phone rings (Call signalling working fine). When the PSTN phone is answered, no media can be heard!!! I did some investigation and here is what is happening: Once the user enters the destination# to dial, The Prepaid/B2BUA does two things: 1) Sends a re-invite to put the UA on hold. 2) Sends an INVITE to the PSTN GW and retrieves the SDP from the response from PSTN GW 3) Sends a second-reinvite to the UA via SER with the SDP info of the PSTNGW 4) SER invokes the mediaproxy (since it is reinvite) and assigns the SAME MEDIAPROXY PORT# as earlier on when the media was flowing fine (SDP has audio port# 35774) 5) SER forwards the re-INVITE TO UA 6) UA responds with a 200 OK and sends 200 OK to SER. SER agains invokes mediaproxy and assigns the SAME MEDIAPROXY PORT# (audio port 35774) and forwards the response to Prepaid (which sends it to PSTNGW). I checked on Cisco PSTNGW that it is creating a session with the mediaproxy (port:35774). BUT NO AUDIO CAN BE HEARD IN ANY DIRECTION. Can anyone please help? Is this a mediaproxy issue that when reinvites are sent and mediaproxy is invoked multiple times, issues arise? I am running the latest mediaproxy version. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Rtpproxy: media not forward
by PREVOT Paul 06 Apr '06

06 Apr '06

Dear all, I have installed rtpproxy from http://www.openser.org/downloads/snapshots/rtpproxy/rtpproxy-cvs-latest.tgz on the same computer as Openser. I started it without options. /var/run/rtpproxy.pid and .sock are created. I suppose a communication is established between OpenSer and Rtpproxy as SDP messages are correctly replaced, it means "Connection Information c" shows rtpproxy IP address and ports. Also both UAs sent RTP media to rtpproxyIP:port described in SDP body, but rtpproxy doesn't forward any RTP packets from one UA to the other. Tcpdump -n udp and host 213.48.xxx.yyy result: 11:48:49.191130 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length 60 11:48:49.250183 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length 60 11:48:49.255407 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length 60 11:48:49.297050 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length 60 11:48:49.297275 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length 60 11:48:49.335890 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length 60 11:48:49.339723 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length 60 11:48:49.393997 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length 60 Have you ever come across that issue? How can I get 207.245.aaa.bbb.35000 > 213.48.xxx.yyy.2237? Does this problem occur because both UAs are in the same local network? (I don't thing so) I would be very thankful if you could give me an idea about it. Thanks a lot Paul

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[Users] Mediaproxy issue
by Jeff Williams 06 Apr '06

06 Apr '06

I am trying to use openser purely as a RTP proxy using mediaproxy. All SIP request are forwarded through to a central server. However, the RTP proxying does not work. The mediaproxy session gets set up and both end points get given the same port on the mediaproxy - is this correct? Both end points forward RTP packets to the mediaproxy but they are not forwarded. These are my log entries for the call and my config is attached: Apr 6 21:33:11 lepus mediaproxy[27804]: command request 89C6D172-1DD2-11B2-BBB5-97F0622A5395(a)192.168.2.3 192.168.2.3:49200:audio 202.74.181.144 sipone.com local 202.74.164.72 remote SJphone/1.60.299a/L=20(SJ=20Labs) info=from:61028679%20@sipone.com,to:1300130710@sipone.com,fromtag:1870227267254482412,totag: Apr 6 21:33:11 lepus mediaproxy[27804]: session 89C6D172-1DD2-11B2-BBB5-97F0622A5395(a)192.168.2.3: started. listening on 202.74.164.44:35038 Apr 6 21:33:11 lepus mediaproxy[27804]: command execution time: 1.54 ms Apr 6 21:33:11 lepus mediaproxy[27804]: command lookup 89C6D172-1DD2-11B2-BBB5-97F0622A5395(a)192.168.2.3 202.74.164.72:20750:audio 202.74.164.72 sipone.com local sipone.com unknown Cisco-SIPGateway/IOS-12.x info=from:61028679%20@sipone.com,to:1300130710@sipone.com,fromtag:1870227267254482412,totag:BDA88B14-B58 Apr 6 21:33:11 lepus mediaproxy[27804]: command execution time: 0.36 ms Apr 6 21:33:19 lepus mediaproxy[27804]: command delete 89C6D172-1DD2-11B2-BBB5-97F0622A5395(a)192.168.2.3 info= Apr 6 21:33:19 lepus mediaproxy[27804]: session 89C6D172-1DD2-11B2-BBB5-97F0622A5395(a)192.168.2.3: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Apr 6 21:33:19 lepus mediaproxy[27804]: session 89C6D172-1DD2-11B2-BBB5-97F0622A5395(a)192.168.2.3: ended. Apr 6 21:33:19 lepus mediaproxy[27804]: command execution time: 0.31 ms I have ethereal captures showing the SIP and RTP packets if they are useful. Can anyone help me? Jeff # $Id: gw-pstn.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=5 fork=yes log_stderror=no port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:pass@localhost/openser" loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/mediaproxy.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/domain.so" #loadmodule "/usr/lib/openser/modules/permissions.so" modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") #modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") #modparam("permissions", "db_mode", 1) #modparam("permissions", "trusted_table", "trusted") modparam("domain", "db_url", "mysql://openserro:pass@localhost/openser") modparam("domain", "db_mode", 1) modparam("usrloc", "db_url", "mysql://openser:pass@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "use_domain", 1) modparam("uri_db", "db_url", "mysql://openserro:pass@localhost/openser") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { add_rr_param(";nat=yes"); record_route(); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { log(">>> loose_route, has_totag, etc"); if (client_nat_test("3")||check_route_param("nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; route(1); } route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; rewritehost("xxx.xxx.xxx.xxx"); route(1); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; rewritehost("xxx.xxx.xxx.xxx"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

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[Serusers] Howto add many users.
by lei_lei＠gmx.net 06 Apr '06

06 Apr '06

Dear All, i have many users to add to my mysql. But if i use e.g. serctl add user1 123456 user1@mydomain ich need to enter the password per hand. So, howto can i do it quickly? Thanks Lei

2 1

[Users] problem replacing from_uri with uac_replace_from
by Raymond Chen 06 Apr '06

06 Apr '06

Dear all, we are trying to replace the from_uri with uac_replace_from function as following, but the actual from_uri is not replaced after uac_replace_from is called as debug log. so when we call load_gw() and next_gw() later in the configuration, the gateway came up is not for the replaced from_uri, it queried with original from_uri. Does anyone know how to completely replace the from_uri from original one. or a avp id that can force the replacement? uac_replace_from("$avp(s:newcallerid)","sip:$avp(s:newcallerid)@xxx.xxx.xxx.xxx:8080"); avp_print(); 0(20433) DEBUG:avpops:print_avp: p=0xf4f18568, flags=3 0(20433) DEBUG: name=<currentcallerid> 0(20433) DEBUG: val_str=<orignalcallerid> thanks Ray

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[Users] What Media Gateway (connected via SS7) do you use
by idont know 06 Apr '06

06 Apr '06

Hello everyone, At first I am sorry that this is slightly off topic. I am doing some research for my company (which does not want to be named therefore the gmail. I am sorry for that) which carrier grade media-Gateways are used by others to connect their SER (or any other SIP-Proxy, - Router, -PBX, etc.) to the PSTN. We are currently looking for something that can be connected via SS7 in Europe, scales good (i.e. starting with 4-E1s, but with no limits to add more) and supports * SIP ;) * G711, G723 and G729 A/B Codecs * Echo Cancellation * VAD * CNG * T.38 * Modem Bypass * DTMF-Transmission using RFC2833 or In band (for G711) and preferably even SIP-INFO I would love to hear which system you use/prefer and what your experiences are ( i.e. would you buy the system you use a second time) and if I forgot something in my requirements that in your eyes is crucial. If you don't want to/can name your company that's fine with me (as I can't either), I just need some opinions. Bye, K

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[Serusers] What Media Gateway (connected via SS7) do you use
by idont know 06 Apr '06

06 Apr '06

Hello everyone, At first I am sorry that this is slightly off topic. I am doing some research for my company (which does not want to be named therefore the gmail. I am sorry for that) which carrier grade media-Gateways are used by others to connect their SER (or any other SIP-Proxy, - Router, -PBX, etc.) to the PSTN. We are currently looking for something that can be connected via SS7 in Europe, scales good (i.e. starting with 4-E1s, but with no limits to add more) and supports * SIP ;) * G711, G723 and G729 A/B Codecs * Echo Cancellation * VAD * CNG * T.38 * Modem Bypass * DTMF-Transmission using RFC2833 or In band (for G711) and preferably even SIP-INFO I would love to hear which system you use/prefer and what your experiences are ( i.e. would you buy the system you use a second time) and if I forgot something in my requirements that in your eyes is crucial. If you don't want to/can name your company that's fine with me (as I can't either), I just need some opinions. Bye, K

1 0

[Serusers] usernames but no pass
by alex wood 06 Apr '06

06 Apr '06

hi i think is got a unique problem. i have a phone system that until the final software package is developed will only allow a user name but no pass and i need to make sure that the server isn't taken for advantage so only a few very long and complex user names will be used. i tried adding a user then going into the mysql and deleting the pass but it seems ser doesn't even get that far if the client doesn't send a pass. so to recap i need a user without a pass but only specified user names to work. also some guy hinted about gateways though I'm not sure he knew what he was on about, so if any of you know that would be good and if their are any documents in ti as well that you could point me to that would be good also. thanks Alex Wood independent contractor

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[Serusers] Warning message
by Suresh Arunachalam 05 Apr '06

05 Apr '06

Hi all, I am using presently ser-0.9.6 and have uac with in the ser. whenever i run my application i get this warning Warning: run_failure_handlers: no UAC support (0, 0) I am not much clear about this warning . can someone give me an idea about this. Thanks&Regards A.Suresh

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Re: [Fwd: Re: [Users] fax support]
by Mike Williams 05 Apr '06

05 Apr '06

I thought that fax over VoIP was unreliable in general? Even with G711, doesn't the packetization break it? Any hints about this are greatly appreciated. ---Mike > Hello, > > On 04/04/06 11:11, Marc Haisenko wrote: > > On Tuesday 04 April 2006 09:49, Christoph Fürstaller wrote: > >> Hi, > >> > >> Is there anyone who uses fax over openser? Or can say something about > >> the reliability of fax over openser? > >> > >> chris... > > > > We're not actively using it but we tested it. It works :-) > > what i can add here is to pay attention of managing re-INVITEs, in the > case you have NATs around. Sending fax uses re-INVITEs to switch to fax > data transmission. > > Cheers, > Daniel > > > C'ya, > > Marc > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -------------------------------------------------------

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[Users] Nathelper and branches
by Andreas M. 05 Apr '06

05 Apr '06

Hello, i have following behavior: when a client is detected behind nat during REGISTER, and i do "fix nated register" i have the public ip address in my location table. but now, when there is an INVITE sent to this user and i append branches, i can see that all invites are sent to the client ip address, that makes no sense. sample config: route[1]{ ....................... append_branch(); avp_pushto("$ruri/username","s:msisdn"); ## ex: 0676123456(a)myproxy.com rewritehostportport("10.0.0.1:5060"); ## my gateway route(2); } route[2]{ if(!t_relay()){ sl_reply_error(); } exit; } with ngrep i can see Initial INVITE to this uri trying................... and the the proxy sends: INVITE sip:initialuri@myproxy.com -> to the ip address that is in the location table (in received field) INVITE sip:0676123456@10.0.0.1:5060 -> also to the ip address that is in the location table in the received field if i make this tests without clients behind nat, all is working fine and one invite is sent to the client and one to the gateway. any ideas ? best regards, Andreas Matzel

2 1

[Users] Openser and Voice Master
by Deepak Singhal 05 Apr '06

05 Apr '06

Hi All, I am facing a problem when terminating calls to Voice Master. The exact problem is : Voice Master is configured for Pin prefix Authentication. I add the gateway prefix in my openser.cfg using prefix("123#") But the Voice Master is rejecting the calls with error 488 Not acceptable here. User dialing number and openser adding prefix to it. But when I send the calls from the userend itself with prefix ie. User is dialing 123#<number> then the calls are going fine. What comes to my mind is that Voice Master is somehow reading the to_uri being sent. Can I somehow add prefix to the to_uri as well ? This is the ngrep trace of the rejected call : U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060 INVITE sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0. Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>. Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0. Via: SIP/2.0/UDP 61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE. From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136. To: <sip:14085264000@XXX.XXX.XXX.XX6>. Contact: <sip:test@61.16.188.4:5060>. Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90. CSeq: 38322 INVITE. Authorization: Digest username="test",realm="XXX.XXX.XXX.XX6",nonce="4433562d730464572c23207912e46 d50281d2136",response="59311eb34dd7ce8b3550064960af1508",uri="sip:1408526400 0(a)XXX.XXX.XXX.XX6". Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-PRO build 1082. Content-Length: 314. . v=0. o=test 3846000 3846000 IN IP4 61.16.188.4. s=X-PRO. c=IN IP4 61.16.188.4. t=0 0. m=audio 8000 RTP/AVP 18 0 8 3 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060 SIP/2.0 100 Trying. From: <sip:test@XXX.XXX.XXX.XX2:5060>;tag=4036363136. To: <sip:14085264000@XXX.XXX.XXX.XX2:5060>;tag=669a9b6321e12fb3fd8a82d7. CSeq: 38322 INVITE. Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90. Via: SIP/2.0/UDP XXX.XXX.XXX.XX6:5060;branch=z9hG4bK5e7a.83627b27.0. Via: SIP/2.0/UDP 61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE. Content-Length: 0. . # U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060 SIP/2.0 488 Not Acceptable Here. Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>. Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0. Via: SIP/2.0/UDP 61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE. From: "test" <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136. To: <sip:14085264000@XXX.XXX.XXX.XX6>. Contact: <sip:test@61.16.188.4:5060>. Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90. CSeq: 38322 INVITE. Authorization: Digest username="test", realm="XXX.XXX.XXX.XX6", nonce="4433562d730464572c23207912e46d50281d2136", response="59311eb34dd7ce8b3550064960af1508", uri="sip:14085264000@XXX.XXX.XXX.XX6". Max-Forwards: 68. Content-Type: application/sdp. User-agent: X-PRO build 1082. Content-Length: 314. . v=0. o=test 3846000 3846000 IN IP4 61.16.188.4. s=X-PRO. c=IN IP4 61.16.188.4. t=0 0. m=audio 8000 RTP/AVP 18 0 8 3 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060 ACK sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0. Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0. From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136. Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90. To: <sip:14085264000@XXX.XXX.XXX.XX6>. CSeq: 38322 ACK. User-Agent: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. . exit 5 received, 0 dropped

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[Users] using AVPs to write into database
by Jayesh Nambiar 05 Apr '06

05 Apr '06

Hi all, I am trying to implement the call forwarding feature in openser. The forwarding part is fine. But I have the follwing scenario: Whenever a user needs to change his forwarding number, he should be able to do so from his end device only. For eg: He first presses 86 and then the 10 digit number to be forwarded. Here the 86 has to be stripped and the 10 digit number should be inserted into the call forward value column of the preferences table. Also if he wants to remove the forwarded number, he can do so by pressing say for eg: 87. on receiving this number openser should delete that value from the table. Is this possible. I tried to do the following, but somehow it does not change the value. if(uri=~"^sip:86[0-9]*@") { if(avp_db_load("$from/username", "s:callfwd")) { #check if call-fwd feature is enabled for the user log(1,"AVP condition returned true"); strip(2); avp_write("$ruri", "s:callfwd"); avp_print(); log(1,"AVP written"); sl_send_reply("200", "OK"); exit; }; }; Is avp_write the proper method or I guess avp_db_store can also help me. The avp_print() function also does not show me anything in the log. Are there any logical mistakes or I have mis-interpreted the syntax of avpops functions. Please help me in thsi regard. Thanks a lot in advance. Jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

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[Users] reseller billing system
by Roberto Pereyra 05 Apr '06

05 Apr '06

Hi Somebody knows a reseller billing system for Openser ? Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: rpereyra(a)lugmen.org.ar For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989

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[Users] External actions
by Manuel Ángel Rubio Jiménez 05 Apr '06

05 Apr '06

Hi, i'm new in OpenSER and i like know, how i can hang up a current call?, i did try it via fifo file, but i haven't found any command to do it. Anybody can help me? Regards. -- Manuel A. Rubio "Bombadil" Usuario de GNU/Linux #323628 acorde a http://counter.li.org/ GPG ID 1C84979D ftp://bosqueviejo.net/pub/bombadil.asc Técnico en Admin. Sistemas Informáticos

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R: [Users] using AVPs to write into database
by D'Addelfio Davide 05 Apr '06

05 Apr '06

Hi guys, I have a similar problem, maybe easier. I'm new in openser so I'd like to have some help. I'd like openser create a table in MySQL db when receive an INVITE, a row with from uri, method and something else, I've look into modules docs but I've not understanding at all...could you help me, please? Thanks for reply Davide ________________________________ Da: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per conto di Jayesh Nambiar Inviato: mercoledì 5 aprile 2006 15.53 A: Bogdan-Andrei Iancu Cc: openser Oggetto: Re: [Users] using AVPs to write into database Hi Bogdan, Thanks for the reply and sorry for sending the earlier mail 4 times as yahoo was not responding peoperly. When I do an avp_write(), why does the avp name get stored in the uuid column? I am not using any uuid as of now. I also tried to add the username of the person who is calling but it again creates two rows, one with the value as username and other row with the value as forwarded number. I thought if there is a way to insert the username also, it wud be easier to check, if the incoming call for that user is to be forwarded or not. I get the following in my usr_preferences table after I dial 86 followed by any number. +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ | uuid | username | domain | attribute | type | value | modified | +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 | 2006-04-05 19:10:49 | +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ The script snippet is as follows: if(uri=~"^sip:86[0-9]*@") { strip(2); avp_write("$ruri", "s:callfwd"); #avp_write("$from/username", "i:999"); #avp_db_store("i:999", "i:/usr_preferences"); avp_db_store("s:callfwd", "s:callfwd/usr_preferences"); sl_send_reply("200", "OK"); exit; }; Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: Hi, you need two steps: 1) write the ruri (after strip) into an AVP - use avp_write() 2) write the avp into the db - use avp_db_store() regards, bogdan Jayesh Nambiar wrote: > Hi all, > I am trying to implement the call forwarding feature in openser. The > forwarding part is fine. But I have the follwing scenario: > Whenever a user needs to change his forwarding number, he should be > able to do so from his end device only. > For eg: He first presses 86 and then the 10 digit number to be > forwarded. Here the 86 has to be stripped and the 10 digit number > should be inserted into the call forward value column of the > preferences table. > Also if he wants to remove the forwarded number, he can do so by > pressing say for eg: 87. on receiving this number openser should > delete that value from the > table. > Is this possible. I tried to do the following, but somehow it does not > change the > value. > if(uri=~"^sip:86[0-9]*@") { > if(avp_db_load("$from/username", "s:callfwd")) { #check > if call-fwd feature is enabled for the user > log(1,"AVP condition returned true"); > strip(2); > avp_write("$ruri", "s:callfwd"); > avp_print(); > log(1,"AVP written"); > sl_send_reply("200", "OK"); > exit; > }; > }; > Is avp_write the proper method or I guess avp_db_store can also help > me. The avp_print() function also does not show me anything in the > log. Are there any > logical mistakes or I have mis-interpreted the syntax of avpops functions. > Please help me in thsi regard. > Thanks a lot in advance. > Jayesh. > > ------------------------------------------------------------------------ > Jiyo cricket on Yahoo! India cricket > > Yahoo! Messenger Mobile > > Stay in touch with your buddies all the time. > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > ________________________________ Jiyo cricket on Yahoo! India cricket <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…> Yahoo! Messenger Mobile <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…> Stay in touch with your buddies all the time.

2 1

[Serusers] Changing the destination port
by Bruno Machado 05 Apr '06

05 Apr '06

Hi all We are trying to install a firewall in our structure. The problem is: the firewall (between X.Y.Z.A and X.Y.Z.B) manipulates the packets, but it changes de SRC port when it sends to the SER (X.Y.Z.B). X.Y.Z.A:5060 ----> X.Y.Z.B:5060 (dump in X.Y.Z.A) X.Y.Z.A:10080 ----> X.Y.Z.B:5060 (doing a dump in X.Y.Z.B) When X.Y.Z.B answer to X.Y.Z.A, it uses the port 10080: X.Y.Z.B:5060 ----> X.Y.Z.A:10080 It break the comunication, because X.Y.Z.A isnt listening at 10080. I need to force the SER to send the messages back to 5060. There are any way to do it? Thanks a lot. Bruno Machado __________________________________________________ Faça ligações para outros computadores com o novo Yahoo! Messenger http://br.beta.messenger.yahoo.com/

2 1

[Users] Use STUN always - is there any downside?
by Barry Flanagan 05 Apr '06

05 Apr '06

Hi, We are using SIPPS as the softphone for our ISP customers, all of whom are on DSL and behind some sort of NAT'ed firewall. SIPPS does not always pick up the need for STUN when it checks the network, and then the user has problems until STUN is enabled. My question is - would there be any downside in us just hard coding the client to use STUN, given that the end user is a home user on DSL? Thanks. -- -Barry Flanagan

2 1

[Serusers] Managing callerid for incoming PSTN
by dashy dude 05 Apr '06

05 Apr '06

Dear All I am new to SER and wanted to figure out how to manage callerid when displayed on a end phone. I have a E1 line terminating on a Cisco Router which we use for inbound/outbound calls. We dial 9 to make a outbound call via this e1. When a incoming call comes, it displays the callerd as presented by my operator. Now, if the user wants to call back, he cant do by the menu of callback since dialing out from pstn needs a digit 9 prefix. I would like to know if I can prefix 9 before forwarding call to PSTN so that the callback from the extension works smoothly. Thanks in advance. Vish __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] Best way to replace digits in dst_uri
by Barry Flanagan 05 Apr '06

05 Apr '06

Hi, I need to rewrite the destination uri to replace "+353" with "0" before passing it on to our PSTN gateway.. What is the cleanest way to do this? Thanks. -- -Barry Flanagan

2 2

[Serusers] Config. SER on SuSE Linux
by Jahangir Mushtaq 05 Apr '06

05 Apr '06

My name is Jahangir I am configuring SER on SuSE Linux.I have some problem to configure it.Any one wants to help me? Jahangir __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] syn_branch
by Simon Morvan 05 Apr '06

05 Apr '06

Just to have the confirmation : When acting as a stateless proxy (not using tm at all), the variable syn_branch have to be 0. syn_branch = 1 is meaningful only in case we use tm ? And the documentation says the default value of syn_branch is 0 but the code initialize the variable to 1. I have to manually set it to 0. -- Simon.

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[Users] ERROR: t_newtran: transaction already in process - using UAC module
by Cameron Beattie 05 Apr '06

05 Apr '06

I am trying to use the UAC module in conjunction with a gateway that requires authentication. I am getting the following when trying to dial a number that gets sent to the gateway: ERROR: t_newtran: transaction already in process Prior to this the following are written to the log file: international PSTN match resetflag(9) host rewritten What seems to be happening is that the t_relay_to_udp is being called and the gateway is contacted but before any reply is received openser is throwing a t_newtran error. My config file is as follows: <snip> route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("60.000.000.000:5060;nat=yes"); # insert IP address } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); return; }; if (uri==myself) { if (method=="ACK") { route(6); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); log(1,"route[2] (Register handler"); if (!search("^Contact: \*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted() && !is_uri_host_local()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (uri=~"^sip:[+|00|011][0-9]*@") { # International PSTN log(1,"international PSTN match"); route(4); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{11}@") { # Domestic(US) PSTN # route(4); return; }; sl_send_reply("404", "User Not Found"); return; }; if (method=="CANCEL") { route(1); return; }; setflag(1); setflag(3); route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- #UAC module routing# t_on_failure("1"); resetflag(9); log(1,"resetflag(9)"); t_relay_to_udp("203.100.100.100","5060"); #PSTN gateway log(1,"host rewritten"); avp_write("i:45", "inv_timeout"); setflag(1); setflag(3); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { log(1,"hit failure_route[1]"); if (t_check_status("401|407")) { log(1,"status is 401 or 407"); if (isflagset(9)) { log(1,"flag9 is set"); t_reply("503","Authentication failed"); return; } if (uac_auth()) { log(1, "called uac_auth"); setflag(9); t_on_failure("1"); append_branch(); t_relay(); } } } The SIP traffic is as follows: U 192.168.0.50:5060 -> 192.168.0.6:5060 INVITE sip:021555222@domain.co.nz SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5 0:5060;rport;branch=z9hG4bK29BECBA8F97C4648A1E027E5DF93BEC0..From: Cameron local Domain <sip:6494466546@domain.co.nz>;tag=3738281931..To: <sip :021555222@domain.co.nz>..Contact: <sip:6494466546@192.168.0.50:5060>.. Call-ID: B1402DE5-6390-4715-A03F-9B1F78AB60CF@192.168.0.50..CSeq: 28000 INV ITE..Proxy-Authorization: Digest username="6494466546",realm="domain.co .nz",nonce="44329b8ecd24c22c48293896310ed7d403738b94",response="014bc155e1a 3554039f6688e9d1e8d42",uri="sip:021555222@domain.co.nz"..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1103m..Conten t-Length: 301....v=0..o=6494466546 766866495 766868168 IN IP4 192.168.0.50. .s=X-Lite..c=IN IP4 192.168.0.50..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 1 01..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpm ap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000. .a=fmtp:101 0-15.. #### U 192.168.0.6:5060 -> 192.168.0.50:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.0.50:5060;rport=5060;branch=z9hG4bK29BECBA8F97C4648A1E027E5DF93BEC0..From : Cameron local Domain <sip:6494466546@domain.co.nz>;tag=3738281931 ..To: <sip:021555222@domain.co.nz>..Call-ID: B1402DE5-6390-4715-A03F-9B 1F78AB60CF@192.168.0.50..CSeq: 28000 INVITE..Server: OpenSer (1.0.1 (i386/l inux))..Content-Length: 0..Warning: 392 192.168.0.6:5060 "Noisy feedback te lls: pid=8125 req_src_ip=192.168.0.50 req_src_port=5060 in_uri=sip:0214462 02(a)domain.co.nz out_uri=sip:021555222@domain.co.nz via_cnt==1".... # U 192.168.0.6:5060 -> 203.184.16.2:5060 INVITE sip:021555222@domain.co.nz SIP/2.0..Record-Route: <sip:60.234.13 7.50:5060;nat=yes;ftag=3738281931;lr=on>..Via: SIP/2.0/UDP 192.168.0.6;bran ch=z9hG4bKbdfc.69423e86.0..Via: SIP/2.0/UDP 192.168.0.50:5060;rport=5060;br anch=z9hG4bK29BECBA8F97C4648A1E027E5DF93BEC0..From: Cameron local domain <sip:6494466546@domain.co.nz>;tag=3738281931..To: <sip:021555222@domain.co.nz>..Contact: <sip:6494466546@192.168.0.50:5060>..Call-ID: B1402D E5-6390-4715-A03F-9B1F78AB60CF@192.168.0.50..CSeq: 28000 INVITE..Max-Forwar ds: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1103m..Co ntent-Length: 301....v=0..o=6494466546 766866495 766868168 IN IP4 192.168.0 .50..s=X-Lite..c=IN IP4 192.168.0.50..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a= rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15.. #### U 192.168.0.6:5060 -> 192.168.0.50:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/ UDP 192.168.0.50:5060;rport=5060;branch=z9hG4bK29BECBA8F97C4648A1E027E5DF93 BEC0..From: Cameron local Domain <sip:6494466546@domain.co.nz>;tag= 3738281931..To: <sip:021555222@domain.co.nz>;tag=329cfeaa6ded039da25ff8 cbb8668bd2.bcdf..Call-ID: B1402DE5-6390-4715-A03F-9B1F78AB60CF(a)192.168.0.50 ..CSeq: 28000 INVITE..Server: OpenSer (1.0.1 (i386/linux))..Content-Length: 0..Warning: 392 192.168.0.6:5060 "Noisy feedback tells: pid=8125 req_src_ ip=192.168.0.50 req_src_port=5060 in_uri=sip:021555222@domain.co.nz out _uri=sip:021555222@domain.co.nz via_cnt==1".... # Any help would be appreciated Cameron

2 1

[Serusers] Contact header during INVITE
by PREVOT Paul 05 Apr '06

05 Apr '06

Hello, I have an issue with the fix_nated_contact() function. OpenSer is supposed to change Contact header, but it only changes it in the INVITE message, but not in the OK response. As a consequence, ACK messages contain sip:test@192.168.1.1:5060 and not the public ip. OpenSer doesn't know how to handle that ACK and forward it to 192.168.1.1:5060. The same issue occurs in CANCEL/487. Changes are done in CANCEL Contact header, but not in 487 reply. No problem for BYE, as OK response doesn't need an ACK I suppose I have something wrong in my cfg file, but I can't find it out. Regards, Paul Openser.Cfg: ...default loadmodules and modparam ... # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER"){ record_route(); force_rport(); fix_nated_register(); fix_nated_contact(); }; #if(search("Contact")){ #setflag(6); #force_rport(); # fix_nated_contact(); #}; if(method!="REGISTER" && method!="ACK") { #record everything besides registers and acks setflag(1); setflag(3); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { #change contact field IP and port by those received really used after NAT rewritting by router #fix_nated_contact(); #force openSer to respond on the port from which the request has been sent insteed of the port #in contact field. force_rport(); # Uncomment this if you want to use digest authentication if (!www_authorize("mydomain", "subscriber")) { www_challenge("mydomain ", "0"); exit; }; setflag(1); setflag(3); fix_nated_register(); save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

2 1

[Users] Registering and rtpproxy/mediaproxy
by Jeff Williams 05 Apr '06

05 Apr '06

We have a commercial SIP server which currently does our registering, low cost routing and customer billing. However, it doesn't handle NATed connections well. To fix this, I wanted to use openser in front of it for customer connections using mediaproxy/rtpproxy. However, I have a problem with this setup. >From what I gather, for the proxy to work properly, the end user need to be registered with the proxy so that it can do nat pings and keep the nat connection alive. But for users to make calls using our commercial server (which is what we want), they must be registered there. Is it possible (or even a good idea) to have a user registered on 2 servers at the same time? How else can we do this? This is what I want to achieve: - from the end user perspective, they register and initiate calls only via the openser proxy. They know nothing of our commercial server. - calls between users is done on the openser proxy - calls to external destinations (i.e. PSTN) go via our commercial server and require that the user is registered there Is this possible? Jeff

3 5

[Users] using AVPs to write into database
by Jayesh Nambiar 05 Apr '06

05 Apr '06

Hi all, I am trying to implement the call forwarding feature in openser. The forwarding part is fine. But I have the follwing scenario: Whenever a user needs to change his forwarding number, he should be able to do so from his end device only. For eg: He first presses 86 and then the 10 digit number to be forwarded. Here the 86 has to be stripped and the 10 digit number should be inserted into the call forward value column of the preferences table. Also if he wants to remove the forwarded number, he can do so by pressing say for eg: 87. on receiving this number openser should delete that value from the table. Is this possible. I tried to do the following, but somehow it does not change the value. if(uri=~"^sip:86[0-9]*@") { if(avp_db_load("$from/username", "s:callfwd")) { #check if call-fwd feature is enabled for the user log(1,"AVP condition returned true"); strip(2); avp_write("$ruri", "s:callfwd"); avp_print(); log(1,"AVP written"); sl_send_reply("200", "OK"); exit; }; }; Is avp_write the proper method or I guess avp_db_store can also help me. The avp_print() function also does not show me anything in the log. Are there any logical mistakes or I have mis-interpreted the syntax of avpops functions. Please help me in thsi regard. Thanks a lot in advance. Jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

1 0

[Users] using AVPs to write into database
by Jayesh Nambiar 05 Apr '06

05 Apr '06

Hi all, I am trying to implement the call forwarding feature in openser. The forwarding part is fine. But I have the follwing scenario: Whenever a user needs to change his forwarding number, he should be able to do so from his end device only. For eg: He first presses 86 and then the 10 digit number to be forwarded. Here the 86 has to be stripped and the 10 digit number should be inserted into the call forward value column of the preferences table. Also if he wants to remove the forwarded number, he can do so by pressing say for eg: 87. on receiving this number openser should delete that value from the table. Is this possible. I tried to do the following, but somehow it does not change the value. if(uri=~"^sip:86[0-9]*@") { if(avp_db_load("$from/username", "s:callfwd")) { #check if call-fwd feature is enabled for the user log(1,"AVP condition returned true"); strip(2); avp_write("$ruri", "s:callfwd"); avp_print(); log(1,"AVP written"); sl_send_reply("200", "OK"); exit; }; }; Is avp_write the proper method or I guess avp_db_store can also help me. The avp_print() function also does not show me anything in the log. Are there any logical mistakes or I have mis-interpreted the syntax of avpops functions. Please help me in thsi regard. Thanks a lot in advance. Jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

1 0

[Users] using AVPs to write into database
by Jayesh Nambiar 05 Apr '06

05 Apr '06

Hi all, I am trying to implement the call forwarding feature in openser. The forwarding part is fine. But I have the follwing scenario: Whenever a user needs to change his forwarding number, he should be able to do so from his end device only. For eg: He first presses 86 and then the 10 digit number to be forwarded. Here the 86 has to be stripped and the 10 digit number should be inserted into the call forward value column of the preferences table. Also if he wants to remove the forwarded number, he can do so by pressing say for eg: 87. on receiving this number openser should delete that value from the table. Is this possible. I tried to do the following, but somehow it does not change the value. if(uri=~"^sip:86[0-9]*@") { if(avp_db_load("$from/username", "s:callfwd")) { #check if call-fwd feature is enabled for the user log(1,"AVP condition returned true"); strip(2); avp_write("$ruri", "s:callfwd"); avp_print(); log(1,"AVP written"); sl_send_reply("200", "OK"); exit; }; }; Is avp_write the proper method or I guess avp_db_store can also help me. The avp_print() function also does not show me anything in the log. Are there any logical mistakes or I have mis-interpreted the syntax of avpops functions. Please help me in thsi regard. Thanks a lot in advance. Jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

1 0

[Users] How to test for domain in sip request in openser.cfg.
by Bas Driessen 05 Apr '06

05 Apr '06

Hello, I would like to test for the domain name in a sip request to decide if an invite is accepted or not. I have the following lines in my openser.cfg file to cater for DNS SRV forwarding: # accept calls from foo.com if ( is_method("INVITE") ) { rewritehost ("sipproxy.foo2.com"); rewriteport (""); } This works well, but I would like to tighten security by only allowing invites forwarded by foo.com. When debugging using xlog calls, the pseudo-variable I am after is : $rd. However $rd can not be used in openser.cfg as follows: if ( is_method("INVITE") && $rd="foo.com" ) Documentation suggests to use core keywords like: if(is_method("INVITE") && from_uri=~".*(a)foo.com") This does not work for my situation and I don't know how I can see the value of 'from_uri', as it can't be expanded into xlog calls. So my kind of overlapping questions are: How can I use pseudo-variables in openser.cfg to test for a certain value? How can I see (debug) the contents of a core keyword? Which keyword should I be using to solve this issue? Has anyone solved a similar issue in the past and if so can you please give some details? Thanks, Bas.

2 2

Re: [Serusers] about the accounting and prepaid problem
by arr 05 Apr '06

05 Apr '06

Dear ALL: I still have this problem !! SIP UA can register in the SER after freeradius checking, but when call forwarded to the b2bua ( b2bua then auth by radius) , follow is the radius log modcall[authorize]: module "sql" returns notfound for request 4 <====but i use the same account and password to register to the SER modcall: leaving group authorize (returns ok) for request 4 auth: No authenticate method (Auth-Type) configuration found for the request: Rejecting the user auth: Failed to validate the user. Please help me follow is in my ser.cfg if (method=="INVITE") { record_route(); forward(b2bua.server,5065); sl_send_reply("181","Forwarding the call to B2BUA"); } ar. ----- Original Message ----- From: Chia-Hung Hsu To: serusers(a)lists.iptel.org Sent: Sunday, March 26, 2006 12:30 AM Subject: Fw: [Serusers] about the accounting and prepaid problem Dear All, I am new in SER and B2BUA, and I have a problem now UA1----SER----B2BUA----Freeradius | | UA2 if i use the configure in b2bua-cfg.xml <Billing_Server> <Address>radius.xxx.com</Address> <Authentication_Port>1813</Authentication_Port> <Accounting_Port>1813</Accounting_Port> <Password>password</Password> </Billing_Server> it will show the log in the radius server, when UA1 called to UA2 rad_recv: Access-Request packet from host 140.110.20.55:1024, id=0, length=68 Authentication-Request sent to a non-authentication port from client b2bua:1024 - ID 0 : IGNORED so , i change the configure in b2bua-cfg.xml <Billing_Server> <Address>radius.xxx.com</Address> <Authentication_Port>1812</Authentication_Port> <Accounting_Port>1813</Accounting_Port> <Password>password</Password> </Billing_Server> when UA1 called to UA2 , caller show the log "Number does not exist Call rejected : 404 not found" Can anyone help me? John Hsu

1 1

[Serusers] how can I set up the SER and Asterisk in two Linux servers?
by Josh Zhou 05 Apr '06

05 Apr '06

Hi, I am new to the VoIP. My plan is to have SER in one Linux server, and have Asterisk in the other Linux server. Can you share your experience with me what I should do with the configuration file in the SER and Asterisk? So that SER and Asterisk can interwork together. Thanks a lot. Josh

2 1

[Serusers] ser 0.9.3 crash
by Tulika Pradhan 04 Apr '06

04 Apr '06

on my machine, ser 0.9.3 crashes as soon as a UA sends REGISTER message on doing gdb ser corefile i get the following when calling 'bt' (gdb) bt #0 qm_malloc (qm=0x8107e00, size=760) at mem/q_malloc.c:290 #1 0x0806ce0e in receive_msg ( buf=0x80d6560 "REGISTER sip:194.255.1.199:5060 SIP/2.0\r\nFrom: <sip:88515356@194.255.1.199>;tag=ccd4c5cb-13c4-2838-9d1b45-6d9e\r\nTo: <sip:88515356@194.255.1.199>\r\nCall-ID: ccd4c5cb-13c4-2838-9d1b45-7a20\r\nCSeq: 1 REGIS"..., len=437, rcv_info=0xbfed0fe0) at receive.c:92 #2 0x0808421c in udp_rcv_loop () at udp_server.c:458 #3 0x0805ceaf in main_loop () at main.c:1032 #4 0x0805e50b in main (argc=2, argv=0xbfed11a4) at main.c:1568 (gdb) print h $9 = 0 please help, note that a similar setup on another machine works just fine. tulika

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[Users] Re: Persistent Data Storage
by Klaus Darilion 04 Apr '06

04 Apr '06

Hi! Please send your emails to the list. This way others can help you too, and others will also benefit. In Openser, all DB task will be done via openser's DB layer. Thus, the modules do not have to take care of the used DB. Under the DB layer, there are the DB drivers: There is at the moment mysql, postgresql and unixodbc. When configuring openser, you only have to use the proper configuration, thus use "postgres://..." isntead of "mysql://...." For setting up the database, use the openser_postgresql.sh script. For setting up openser in general, please refer to the Gettign Started Tutorial on onsip.org regards klaus voiceoverip wrote: > Hi Klaus, > > forgive me for being a nuisance:) I've tried google and Openser manuals, > haven't been able to find any revelant Openser setup guide with > PostgreSQL, would be glad if you can point me to any online reference(s). > > Also, I assume that by native, you meant the client interface to the > underlying DB platform. Therefore, Postgres support would still be > essentially modular? > > Thanks for your patience. > > Regards > > Klaus Darilion writes: > > voiceoverip wrote: > >> Hi Klaus, > >> thanks for your advice, will check it out. Can i say that MySQL support > >> is native? Mind if i ask what major difference(s) between modular and > >> native support? > > > > mysql and postgres have native support. Both works well. I've never > tried > > the unixodbc yet. > > > > regards > > klaus > > > >> Regards > >> voiceoverip > >> Klaus Darilion writes: > >> > voiceoverip wrote: > >> >> Hi, > >> >> I am new to OpenSER. > >> >> As noted in the installation notes(Persistent Data Storage) - "you > >> need > >> >> to make sure MySQL is installed on your box". Guess other > backends(in > >> >> particular PostgreSQL) are not supported. If true, wonder if > there are > >> >> plans to support other backends. Thanks. > >> > > >> > Take a look at the modules directory. There is already postgresql and > >> > unixodbc. > >> > > >> > There is also native oralce support in ser's experimental tree. > >> > > >> > regards > >> > klaus > >> ________________________________ > >> Free POP3 Email from www.gawab.com > >> Sign up NOW and get your account @gawab.com!! > > > ________________________________ > Free POP3 Email from www.gawab.com <http://www.gawab.com> > Sign up NOW and get your account @gawab.com!!

2 1

RE: [Serusers] Ser as a presence server with other SIP proxy asregistrar and rtp proxy
by İlker Aktuna (Koç.net) 04 Apr '06

04 Apr '06

Hi again, I think I've found the problem. The client doesn't receive the packet from server because of network problems. So I had to change the SIP port of my SIP Proxy to 5061 Now , following forward command does not work (SER still sends to 5060) , why ? if (!t_forward_nonack("193.243.202.97", 5061)) { log(1, "forward failed\n"); t_reply("500", "Forward failed"); } Thanks, ilker ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, March 31, 2006 3:21 PM To: Vaclav Kubart Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Ser as a presence server with other SIP proxy asregistrar and rtp proxy Hi Vaclav, Thanks for the information. I changed my configuration for REGISTER messages and they were forwarded to my SIP proxy. But the clien't can't connect. I think that's because of authentication requirement on my SIP proxy. When I make the client connect directly to my SIP proxy I see the following SIP header on it: 12:09:22.967260: SIP: 193.243.202.97:5060 << 85.105.102.167:8359 payload: 792 bytes REGISTER sip:193.243.202.97 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.6:8359;branch=z9hG4bK-d87543-c2540260b6061b1e-1--d87543-;rport Max-Forwards: 70 Contact: <sip:200000900568888888@192.168.2.6:8359;rinstance=c314db726308b27a> To: "900568888888"<sip:200000900568888888@193.243.202.97> From: "900568888888"<sip:200000900568888888@193.243.202.97>;tag=f72f1c2c Call-ID: 2748934b5d115c55@TkVUV09SS1lPTi5jb21wYW55MS50c3Q. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3014w stamp 26359 Authorization: Digest username="200000900568888888",realm="NcxSip",nonce="48058627",uri="sip:193.243.202.97",response="92be0838ca4f794d23d4bb3d34e12aa9",algorithm=MD5 Content-Length: 0 But if I make the client connect to SER and then forward the REGISTER message to my SIP proxy, I see the following SIP header arriving: 12:16:42.263635: SIP: 193.243.202.97:5060 << 193.243.207.107:10003 payload: 716 bytes REGISTER sip:193.243.202.97 SIP/2.0 Via: SIP/2.0/UDP 10.100.100.15;branch=z9hG4bKcee1.8c792d45.0 Via: SIP/2.0/UDP 192.168.2.6:8359;received=85.105.102.167;branch=z9hG4bK-d87543-aa1ee14dcf543535-1--d87543-;rport=10002 Max-Forwards: 16 Contact: <sip:200000900568888888@192.168.2.6:8359;rinstance=2c28e04a9c62f011> To: "900568888888"<sip:200000900568888888@193.243.202.97> From: "900568888888"<sip:200000900568888888@193.243.202.97>;tag=d37fe827 Call-ID: 6d7923631c258f31@TkVUV09SS1lPTi5jb21wYW55MS50c3Q. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3014w stamp 26359 Content-Length: 0 It seems that when the SIP message is forwarded through SER, the Authorization information is missing. How can I make SER to forward the message without modifying the message ? Also how can I modify user and domain information for the forwarded messages ? Thanks, ilker -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Friday, March 31, 2006 8:37 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Ser as a presence server with other SIP proxy as registrar and rtp proxy Hi, your solution is possible - separated presence server from other SIP communication. For forwarding you need cfg like this: if (method=="...") { # here put method name which should be forwarded if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; if (!t_forward_nonack("machine.domain.com", 5060)) { log(1, "forward failed\n"); t_reply("500", "Forward failed"); } break; } I'm using SER as proxy and other SER as presence server. Proxy forwards presence related messages to presence server, it looks like: C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi again, I think I've found the problem. The client doesn't receive the packet from server because of network problems. So I had to change the SIP port of my SIP Proxy to 5061 Now , following forward command does not work (SER still sends to 5060) , why ? if (!t_forward_nonack("193.243.202.97", 5061)) { log(1, "forward failed\n"); t_reply("500", "Forward failed"); } Thanks, ilker ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, March 31, 2006 3:21 PM To: Vaclav Kubart Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Ser as a presence server with other SIP proxy asregistrar and rtp proxy Hi Vaclav, Thanks for the information. I changed my configuration for REGISTER messages and they were forwarded to my SIP proxy. But the clien't can't connect. I think that's because of authentication requirement on my SIP proxy. When I make the client connect directly to my SIP proxy I see the following SIP header on it: 12:09:22.967260: SIP: 193.243.202.97:5060 << 85.105.102.167:8359 payload: 792 bytes REGISTER sip:193.243.202.97 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.6:8359;branch=z9hG4bK-d87543-c2540260b6061b1e-1--d87543-;rport Max-Forwards: 70 Contact: <sip:200000900568888888@192.168.2.6:8359;rinstance=c314db726308b27a> To: "900568888888"<sip:200000900568888888@193.243.202.97> From: "900568888888"<sip:200000900568888888@193.243.202.97>;tag=f72f1c2c Call-ID: 2748934b5d115c55@TkVUV09SS1lPTi5jb21wYW55MS50c3Q. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3014w stamp 26359 Authorization: Digest username="200000900568888888",realm="NcxSip",nonce="48058627",uri="sip:193.243.202.97",response="92be0838ca4f794d23d4bb3d34e12aa9",algorithm=MD5 Content-Length: 0 But if I make the client connect to SER and then forward the REGISTER message to my SIP proxy, I see the following SIP header arriving: 12:16:42.263635: SIP: 193.243.202.97:5060 << 193.243.207.107:10003 payload: 716 bytes REGISTER sip:193.243.202.97 SIP/2.0 Via: SIP/2.0/UDP 10.100.100.15;branch=z9hG4bKcee1.8c792d45.0 Via: SIP/2.0/UDP 192.168.2.6:8359;received=85.105.102.167;branch=z9hG4bK-d87543-aa1ee14dcf543535-1--d87543-;rport=10002 Max-Forwards: 16 Contact: <sip:200000900568888888@192.168.2.6:8359;rinstance=2c28e04a9c62f011> To: "900568888888"<sip:200000900568888888@193.243.202.97> From: "900568888888"<sip:200000900568888888@193.243.202.97>;tag=d37fe827 Call-ID: 6d7923631c258f31@TkVUV09SS1lPTi5jb21wYW55MS50c3Q. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3014w stamp 26359 Content-Length: 0 It seems that when the SIP message is forwarded through SER, the Authorization information is missing. How can I make SER to forward the message without modifying the message ? Also how can I modify user and domain information for the forwarded messages ? Thanks, ilker -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Friday, March 31, 2006 8:37 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Ser as a presence server with other SIP proxy as registrar and rtp proxy Hi, your solution is possible - separated presence server from other SIP communication. For forwarding you need cfg like this: if (method=="...") { # here put method name which should be forwarded if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; if (!t_forward_nonack("machine.domain.com", 5060)) { log(1, "forward failed\n"); t_reply("500", "Forward failed"); } break; } I'm using SER as proxy and other SER as presence server. Proxy forwards presence related messages to presence server, it looks like: <http://387555.sigclick.mailinfo.com/sigclick/03090F04/06044907/014D0A101813…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] Behavior of next_gw
by Florian Fainelli 04 Apr '06

04 Apr '06

Hello, I would like to know what is the default behavior when using next_gw(). Here is basic example : A call is coming to gw 1, gateway no 1 rejects it, the next gateway is gw2, is this call forwarded to gw2 and so on ... ? Thank you very much in advance for your answer. Florian

2 1

[Serusers] Limiting number of calls
by Waldo Rubinstein 04 Apr '06

04 Apr '06

Is there a way to restrict the number of simultaneous calls for a particular sip client? For example, if I sell a service where each line can have at most 4 simultaneous call voice paths and there are two incoming calls and two separate outgoing calls, how can I make it so that whenever a new calls comes in or a user tries to make another call, they get fast busy or just busy? Thanks, Waldo

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[Fwd: Re: [Users] fax support]
by Christoph Fürstaller 04 Apr '06

04 Apr '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Is there anyone who uses fax over openser? Or can say something about the reliability of fax over openser? chris... Hi, only end devices need to support this feature. Nothing to be set on openser side. Ramona unplug wrote: >Hi, > Does openser support fax? How? >unplug > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEMiT6R0exH8dhr/YRAu8rAJ0cNvJRRqSHycU+dhMZvuJeVK+JmACg0Uox rk/ECrupb92VR2+wBmkovmM= =0pKc -----END PGP SIGNATURE-----

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[Serusers] Anyone have config samples for asterisk-b2bua/SER?
by sip 04 Apr '06

04 Apr '06

We're looking into Asterisk-b2bua with SER to handle prepaid stuff, but we're running into the snag that, well, there's pretty much absolutely NO documentation for asterisk-b2bua. It's a few perl modules, a config file, and an INSTALL file that consists of explaining how to download the source and patch asterisk to use the program. That's it. It mentions radius, and it LOOKS like radius needs to be running on the same box as the asterisk-b2bua setup. I suppose that will work, as I can then point asterisk to do its accounting/auth on the SER box where I have radius running for auth. I've seen mention that asterisk-b2bua doesn't handle digest authentication (although it has an auth server in its config file), but I don't even know what it would be doing authentication FOR at that point and, since SER uses digest, I don't know how the two would mesh. Other than what I've been able to figure out from looking at the code (which is, granted, not a whole lot), there's nothing even remotely resembling docs (oh, my kingdom for the ability to force coders to hire technical writers). I was kind of hoping someone might have a sample config for use with SER... or be able to point me in the direction of one. N.

2 1

[Users] fax support
by unplug 04 Apr '06

04 Apr '06

Hi, Does openser support fax? How? unplug

2 1

[Serusers] I want to know if someones can help us to building a meetme conference?
by zhaomin 04 Apr '06

04 Apr '06

I have been used SER and Asterisk ,I know asterisk have a function of meetme conference,but I don't know how to use it .Anyone know how to use and configure or administrat it Thankful Zhaomin

1 0

[Users] Mediaproxy and Prepaid/B2BUA interop issue!
by Dave 04 Apr '06

04 Apr '06

Hello,I have the following setup: UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW | IVR Nat'd user calls a number that is forwarded (t_relay) to Prepaid and the user is prompted for PIN#/Destination Phone# and all works fine. Mediaproxy is invoked and a mediaproxy port# (35774) is assigned during the whole process. Once the user enters the destination number, the Prepaid forwards the call to the PSTNGW and the PSTN phone rings (Call signalling working fine). When the PSTN phone is answered, no media can be heard!!! I did some investigation and here is what is happening: Once the user enters the destination# to dial, The Prepaid/B2BUA does two things: 1) Sends a re-invite to put the UA on hold. 2) Sends an INVITE to the PSTN GW and retrieves the SDP from the response from PSTN GW 3) Sends a second-reinvite to the UA via SER with the SDP info of the PSTNGW 4) SER invokes the mediaproxy (since it is reinvite) and assigns the SAME MEDIAPROXY PORT# as earlier on when the media was flowing fine (SDP has audio port# 35774) 5) SER forwards the re-INVITE TO UA 6) UA responds with a 200 OK and sends 200 OK to SER. SER agains invokes mediaproxy and assigns the SAME MEDIAPROXY PORT# (audio port 35774) and forwards the response to Prepaid (which sends it to PSTNGW). I checked on Cisco PSTNGW that it is creating a session with the mediaproxy (port:35774). BUT NO AUDIO CAN BE HEARD IN ANY DIRECTION. Can anyone please help? Is this a mediaproxy issue that when reinvites are sent and mediaproxy is invoked multiple times, issues arise? I am running the latest mediaproxy version. Thanks Dave --------------------------------- Make Yahoo! Canada your Homepage Yahoo! Canada Homepage

1 0

[Serusers] SER, Mediaproxy and Prepaid interop issue
by Dave 04 Apr '06

04 Apr '06

Hello,I have the following setup: UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW | IVR Nat'd user calls a number that is forwarded (t_relay) to Prepaid and the user is prompted for PIN#/Destination Phone# and all works fine. Mediaproxy is invoked and a mediaproxy port# (35774) is assigned during the whole process. Once the user enters the destination number, the Prepaid forwards the call to the PSTNGW and the PSTN phone rings (Call signalling working fine). When the PSTN phone is answered, no media can be heard!!! I did some investigation and here is what is happening: Once the user enters the destination# to dial, The Prepaid/B2BUA does two things: 1) Sends a re-invite to put the UA on hold. 2) Sends an INVITE to the PSTN GW and retrieves the SDP from the response from PSTN GW 3) Sends a second-reinvite to the UA via SER with the SDP info of the PSTNGW 4) SER invokes the mediaproxy (since it is reinvite) and assigns the SAME MEDIAPROXY PORT# as earlier on when the media was flowing fine (SDP has audio port# 35774) 5) SER forwards the re-INVITE TO UA 6) UA responds with a 200 OK and sends 200 OK to SER. SER agains invokes mediaproxy and assigns the SAME MEDIAPROXY PORT# (audio port 35774) and forwards the response to Prepaid (which sends it to PSTNGW). I checked on Cisco PSTNGW that it is creating a session with the mediaproxy (port:35774). BUT NO AUDIO CAN BE HEARD IN ANY DIRECTION. Can anyone please help? Is this a mediaproxy issue that when reinvites are sent and mediaproxy is invoked multiple times, issues arise? I am running ser-0.9.6 and the latest mediaproxy version. Thanks Dave __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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R: R: R: [Users] export SIP_Domain
by D'Addelfio Davide 03 Apr '06

03 Apr '06

Thanks. Do you think is possible, instead to work with script, to work over source code and change it to let it do what I want? Any thought? -----Messaggio originale----- Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Inviato: giovedì 30 marzo 2006 15.56 A: D'Addelfio Davide Cc: users(a)openser.org Oggetto: Re: R: R: [Users] export SIP_Domain On 03/29/06 12:50, D'Addelfio Davide wrote: > Ok thanks, will take a look! > > Now that I've installed and running openser, I have to work over it. Do you know if is possibile to configure openser with some trigger that allows it to switch incoming message (INVITE, REQUEST, etc...) towards another element, like an application server? Do they exists? Where can I find it? > do you want to route certain request types to different hosts? Like if request is INVITE, send it to an application server? If so, then you can do: if(is _method("INVITE")) { t_relay("udp:app_srv_ip:port"); exit; } This example works with development version of openser, for versions 1.0.x below, check the documentation of tm module which is posted on project's web site (http://openser.org). You have to use the textops module as well. Cheers, Daniel > Thanks for help > > -----Messaggio originale----- > Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Inviato: mercoledì 29 marzo 2006 11.47 > A: D'Addelfio Davide > Cc: users(a)openser.org > Oggetto: Re: R: [Users] export SIP_Domain > > On 03/29/06 11:56, D'Addelfio Davide wrote: > >> Thanks for reply...it works very good!!! >> >> Still not understand which different between SER and OpenSER, could anyone explain me? >> >> > openser started from the same code base as ser v0.9.x, and that was > openser v0.9.x . Since then a lot of features has be added, new stable > release was made as 1.0.0 in October last year. For more details see: > > http://openser.org > http://openser.org/diffs-0.8.14.php > http://openser.org/diffs-0.9.0.php > http://openser.org/release-1.0.0.php > > Cheers, > Daniel > > >> Thanks >> >> -----Messaggio originale----- >> Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] >> Inviato: martedì 28 marzo 2006 11.55 >> A: D'Addelfio Davide >> Cc: users(a)openser.org >> Oggetto: Re: [Users] export SIP_Domain >> >> Hello, >> >> On 03/28/06 11:59, D'Addelfio Davide wrote: >> >> >>> Hi to all, i'm totally new in openser and i'd like to enjoy it... >>> >>> I've installed 1.0.1 version on my debian and seems it work fine with >>> mysql database. >>> >>> Just a stupid question: I miss to export SIP_DOMAIN cause I can't >>> understand what name I have to give...the default is openser.org...and >>> I've seen on install file that I have to change it...could someone >>> explain me what I have to do, please? >>> >>> >>> >> the SIP_DOMAIN has to be your domain or IP address: >> >> export SIP_DOMAIN="mydomain.com" >> >> Cheers, >> Daniel >> >> >> >>> Thanks in advance >>> >>> Davide >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > >

2 1

[Serusers] What is a function that it can handle stateless relay?
by Ren-Huang Liou 03 Apr '06

03 Apr '06

Hi, I know that t_relay() can use for stateful transaction relay. What is a function that it can handle stateless relay? Does it exist? Thanks in advance. Best regards, Ren-Huang Liou

1 0

[Serusers] adding a user without a pass
by alex wood 03 Apr '06

03 Apr '06

i havle little to none mysql skills so enless you can right a guide on how to do it that road is imposible to me :P sorry to say im on centos 4.3 server ed so no gui and no mysql administrator befor i start installing a lamp on my box to get phpmyadmin running i was wondering if there was a magic command that couold helkp me out like instaead of serctl add someguy somepass some(a)thing.com i could do serctl add someguy null some(a)thing.com or some combo of " or ' or even - to trick mysql to not put anything in that colum but keep ser happy :) thanks for reading this far :) regards Alex Wood

4 3

[Serusers] SER PRACK
by Razvan Nemesiu 03 Apr '06

03 Apr '06

Dear all, Does anyone know if SER (0.8.14) supports PRACK messages ? I want to know if it forwards them.I want to add in my configuration if(method==PRACK) {..} and I want to know if I can do that and if SER allows this configuration line. Best regards, Razvan

1 0

[Serusers] RTCP compliant Phone
by Aisling 03 Apr '06

03 Apr '06

Hello, This question isn't directly related to SER so I apologise, however I am finding it difficult to find a windows softphone that is fully RTCP compliant (according to RFC 1889, 3550). I use a SER platform. I was using XLite and discovered that they have implemented the RTCP stack incorrectly. I subsequently tried a copy of Eyebeam and unfortunately there are errors in that implementation also. SJPhone doesn't seem to send RTCP values and with Windows Messenger some of the RTCP fields are consistently set to 0 which is most definitely incorrect. If anyone can advise on this I would be very grateful. Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] distinctive ring
by Ramin NIkaeen 03 Apr '06

03 Apr '06

Hi Everyone, Can anyone point me in the right direction as to how to implement distinctive ringing using SER?! Has anyone attempted it at all?! I would very much appreciate your help ramin

4 3

[Users] Cisco phone behind ADSL (with NAT)
by Joao Pereira 02 Apr '06

02 Apr '06

Hello to all Did you ever made it to put a Cisco phone (Cisco IP phone 7940 with SIP) behind ADSL (and NAT) and make calls?? Im trying to do it, and with X-Lite I can place calls, but with the Cisco phone I cant... :( I already putted the NAT= yes in the Cisco Phone. Is there any way of doing this ? Thanks Joao Pereira www.fccn.pt

3 4

[Serusers] distinctive ring
by Adrian Georgescu 02 Apr '06

02 Apr '06

Look for Alert-Info header. Adrian >>>>> Hi Everyone, Can anyone point me in the right direction as to how to implement distinctive ringing using SER?! Has anyone attempted it at all?! I would very much appreciate your help ramin

1 0

[Serusers] SER Billing
by Adrian Georgescu 02 Apr '06

02 Apr '06

Try CDRTool it is now open source http://cdrtool.ag-projects.com Adrian >>>>>>>>>>>>>> Hi all! I am a newcomer to this list Anyone who can point me in right direction to find a billing software that works with SER? All suggestions are welcome, open source or commercial Regards ------------------------------------ Talky Communications Anders Svensson Manager anders(a)talkycom.com Sjörrödsvägen 1 S-280 23 Hästveda tel: +46 40 608 2250 mobile: +46 703 17 13 06 ------------------------------------

1 0

[Serusers] SER Billing
by Anders Svensson 02 Apr '06

02 Apr '06

Hi all! I am a newcomer to this list Anyone who can point me in right direction to find a billing software that works with SER? All suggestions are welcome, open source or commercial Regards ------------------------------------ Talky Communications Anders Svensson Manager anders(a)talkycom.com Sjörrödsvägen 1 S-280 23 Hästveda tel: +46 40 608 2250 mobile: +46 703 17 13 06 ------------------------------------

1 0

[Serusers] Call forwarding(at no answer) doesn't work in failure route.
by Yuhai Qin - Snow Line 02 Apr '06

02 Apr '06

I have been jailed by this script for almost one week. I refered all archives available(including Onsip's Getting Started) and simply couldn't make the Call Forwarding(at no answer) run properly. In route[1], I used: if (isflagset(28)){ avp_pushto("$ruri", "s:fwdnoanswer"); append_branch(); lookup("location"); xlog("L_INFO", "No answer, relaying after lookup. method <%rm> From <%fu> To <%tu> Contact <%ct> r-uri <%ru>\n"); }; t_relay(); In failure _route[1], I used: if (isflagset(27) && t_check_status("408")) { setflag(28); revert_uri(); route(1); }; I think this is quite simple and straightforward,right? However, after I did revert_uri, avp_pushto, append_branch and lookup, t_relay never could relay the INVITE message to the new branch(I knew this because I checked with ngrep and found no any INVITE message for the new branch). Anyone can help? It's urgent! Thank you so much! Jerry. P.S., here is my original ser.cfg. Just for your reference: ============================================================================= debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) listen=xx.xx.xx.xx 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" fifo_db_url="mysql://ser:heslo@localhost/ser" alias="sip.xx.com" alias="xx.xx.xx.xx" alias="xx.com" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) 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 modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 30 ) modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("uri_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1 ) modparam("acc", "db_url","mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 1 ) modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") # ------------------------- request routing logic ------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method == "REGISTER" || ! search("^Record-Route:")) { 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 }; setflag(1); # used for acc. if (method=="INVITE") log(1, "INVITE message received\n"); if (method=="ACK") log(1, "ACK message received\n"); if (method=="BYE") log(1, "BYE message received\n"); if (method=="CANCEL") log(1, "CANCEL message received\n"); if (!method=="REGISTER") record_route(); if (loose_route()) { route(1); break; }; if (uri!=myself) { route(1); break; }; if (method=="REGISTER") { save("location"); break; }; if (method=="INVITE") { if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { setflag(27); log(1, "dbg: FWDBUSY/FWDNOANSWER recorded!\n"); lookup("location"); }; }; t_on_failure("1"); route(1); } route[1] { if (isflagset(6)) force_rtp_proxy(); if (isflagset(28)){ avp_pushto("$ruri", "s:fwdnoanswer"); append_branch(); lookup("location"); xlog("L_INFO", "No answer, relaying after lookup. method <%rm> From <%fu> To <%tu> Contact <%ct> r-uri <%ru>\n"); }; t_relay(); } failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(27) && t_check_status("408")) { revert_uri(); setflag(28); route(1); }; }

1 0

[Serusers] to many hops
by alex wood 01 Apr '06

01 Apr '06

i keep getting to many hops when i enable the mysql feature in ser useing http://www.iptel.org/ser/doc/ser-howto/ser-Howto.html and the latest version of ser im on cent os 4.3 server ed i think its 4.3 i think thats all the info you need. thanks Alex Wood

1 0

[Serusers] Review ser.cfg - SER NAT RTPProxy PSTN Gateway
by Ronald Voermans 01 Apr '06

01 Apr '06

Hello, I have a question regarding my ser.cfg. First let me explain our situation. I have a SER server (actually openser) behind a NAT firewall (Cisco PIX, which is doing SIP ALG). The SER server has ip address 10.254.254.1 which is translated by de NAT Firewall to a public IP address. Behind the SER we have a connect with a telco who provides us a SIP PSTN gateway. It's IP address is 10.166.38.108 (so, also private). Remote clients connect to our SER via the public IP address of the SER. For this to work, I use RTPProxy for all sessions. So I do NO client_nat tests or whatsoever. Most of the time, all calls go well. However sometimes when a UAC retransmits INVITE requests, the retransmitted responses display the wrong proxy-address (for RTPProxy). For example, when the UAC does 4 INVITE request (so 1 request, and 3 retransmissions), 3 of the responses have SDP with 10.254.254.1 in their body. The last one doesn't change the SDP address, but has 10.166.38.108 in its body. What could be causing this? Clients obviously don't get audio in their conversation. Next, since I wrote the ser.cfg by my own, it would be nice if some of you could review my ser.cfg and, if needed, give me some suggestions etc. Thank you all very much in advance, My ser.cfg: ######################################################################## ###################################### debug=3 fork=yes log_stderror=no #debug=4 #fork=no #log_stderror=yes listen=10.254.254.1 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://XXX:XXX@localhost/ser" fifo_mode=0666 loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/exec.so" loadmodule "/usr/local/lib/openser/modules/group.so" modparam("auth_db|permissions|uri_db|usrloc|acc|domain|group", "db_url", "mysql://XXX:XXX@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (search("User-Agent: Grandstream")) { replace("UPDATE,", ""); } if (search("Server: Cisco ATA.*")) { replace(", UPDATE", ""); } if (search("User-Agent: Cisco ATA.*")) { replace(", UPDATE", ""); } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { setflag(6); force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || src_ip=="10.166.38.108") { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; exit; }; if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="ACK") { # Added 28-3-06: I won't send a reply for ACK! route(1); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); setflag(1); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!src_ip=="10.166.38.108") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; }; consume_credentials(); }; setflag(6); lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { if (uri=~"^sip:0[0-9]*@") { # PSTN route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.166.38.108"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(4); route(1); } onreply_route[1] { if (search("Server: Cisco ATA.*")) { replace(", UPDATE", ""); } if (search("User-Agent: Grandstream")) { replace("UPDATE,", ""); } if (isflagset(6) && (status=~"180|183|2[0-9][0-9]")) { if (search("application/sdp")) { force_rtp_proxy(); }; }; fix_nated_contact(); } ######################################################################## ################ Regards, Ronald

1 0

[Serusers] HELP NEEDED
by Info 01 Apr '06

01 Apr '06

Hi, I am new to SER as our company previous administrator who looked after our SIP server has left the company and left no documentation. Our server got restarted du to power failure and it is unable to start ser. This is what I am doing to start SER and I am getting following error. /usr/local/sbin/ser -f /usr/local/etc/ser/ser_091105-noenum-full.cfg and this is what I am getting in /var/log/messages file: 0(2772) parse error (81,38-39): unknown command, missing loadmodule? 0(2772) parse error (82,2-3): syntax error 0(2772) parse error (82,2-3): bad command 0(2772) parse error (82,4-8): ERROR: bad config file (4 errors) What I am trying to find out what is (81,38-39)? is it line 81 column 38-39 or else also how can I fix this problem. Please help me so I could get our phone system working again. Regards -- ( Connecting You To The World ) SMS gigo to 82280 for our Contact Details

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