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[Serusers] Fwd: Kphone, TLS and SER ... bug, and patch
by Cesc Santasusana 18 Mar '05

18 Mar '05

Sorry guys ... my fault ... the previous text was a screw up ... I send it now with proper windows formatting ... Cesc >>> Cesc Santasusana 03/18/05 02:36PM >>> Hi, With the advent of TLS support for SER, i was testing it with KPhone (TLS being used between SER proxies only, Kphone-to-SER via UDP). Turned out that kphone would not respond to any message (it would receive them, but plain ignore them). The problem: TLS as transport mode in the VIA headers was not supported, thus, the SIP messages were considered not valid. I am attaching a patch to correct this for kphone 4.1.0, but i guess it can be applied with little modifications to earlier versions (4.0.3 to 4.0.5) but i am not sure. Regards, Cesc PATCH STARTS HERE: Index: dissipate2/sipclient.cpp =================================================================== --- dissipate2/sipclient.cpp (revision 1) +++ dissipate2/sipclient.cpp (working copy) @@ -1034,6 +1034,9 @@ printf( "SipClient: Sending TCP Response\n" ); outsocket = new TCPMessageSocket; break; + case SipVia::TLS: + printf( "SipClient: TLS in top via, not supported (full TLS support not implemented)\n" ); + break; case SipVia::BadTransport: printf( "SipClient: Bad transport on incoming Via\n" ); break; @@ -1232,6 +1235,9 @@ printf( "SipClient: Sending TCP Response\n" ); outsocket = new TCPMessageSocket; break; + case SipVia::TLS: + printf( "SipClient: TLS in top via, not supported (full TLS support not implemented)\n" ); + break; case SipVia::BadTransport: printf( "SipClient: Bad transport on incoming Via\n" ); break; Index: dissipate2/sipvia.h =================================================================== --- dissipate2/sipvia.h (revision 1) +++ dissipate2/sipvia.h (working copy) @@ -49,6 +49,7 @@ enum Transport { UDP, TCP, + TLS, BadTransport }; /** Index: dissipate2/sipvia.cpp =================================================================== --- dissipate2/sipvia.cpp (revision 1) +++ dissipate2/sipvia.cpp (working copy) @@ -37,6 +37,7 @@ switch ( t ) { case UDP: return "UDP"; case TCP: return "TCP"; + case TLS: return "TLS"; case BadTransport: return "BAD"; } return QString::null; @@ -46,6 +47,7 @@ { if ( t.compare( getTransportString( UDP ) ) == 0 ) { return UDP; } if ( t.compare( getTransportString( TCP ) ) == 0 ) { return TCP; }; + if ( t.compare( getTransportString( TLS ) ) == 0 ) { return TLS; }; return BadTransport; } PATCH ENDS HERE Unclassified

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[Serusers] SER x Nextone problem
by Rodrigo P. Telles 18 Mar '05

18 Mar '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi guys, I've been experiencing some troubles with carrier's that are using Nextone (www.nextone.com). I could place calls through Nextone but no longer then 1 minute, between 48 and 53 seconds the call is dropped with a BYE sent by Nextone. I've been using another carriers using Cisco and Quintum without problems. Does some one experimenting this problem? I'm using SER 0.8.14. Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> IVOZ # 1009 Project Manager Devel-IT - http://www.devel.it Bestcom Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCOv2MiLK8unYgEMQRAhlpAJ0d13DsSvWSRlXQc++aLkJC+TdSRgCfbOxe dlm/fu6iI8XbJfoAC4CRxtU= =VxdG -----END PGP SIGNATURE-----

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[Serusers] DUV 1000 problems w/ www_authorize
by Pavel Siderov - Hostmates 18 Mar '05

18 Mar '05

Hi guys, I have a problem w/ the DUV 1000 ( http://corp.deltathree.com/productsandservices/dialup.html ) hardware device. I'm using www_authorize in my ser.cfg on every call forward (pstn) to prevent other sip proxies to forwad traffic to me: if (uri=~"^sip:00") { log(0, "\n\n************ B2BUA ***********\n"); if((!src_ip=="12.34.56.78")||(!src_ip=="1.2.3.4")){ # if (!proxy_authorize("12.34.56.78", "subscriber")) { # proxy_challenge("12.34.56.78", "1"); # }; if (!www_authorize("12.34.56.78", "subscriber")) { www_challenge("12.34.56.78", "1"); break; }; }; strip(0); rewritehostport("1.2.3.4:11234"); append_branch("1.2.3.4"); t_relay(); t_relay_to_udp("1.2.3.4", "11234"); append_hf("P-hint: VoipSwitch GATEWAY\r\n"); log(0, "********************************************\n\n\n"); break; }; When I try to call 00xxxxxx the DUV device doesn't authorize to the ser and the call fails. The same happens when proxy_authorize is used. Here is the sniffer log: # U 34.56.87.12:5060 -> 12.34.56.78:5060 INVITE sip:00359123456@12.34.56.78 SIP/2.0. Via: SIP/2.0/UDP 34.56.87.12:5060. From: 6789 <sip:6789@12.34.56.78>;tag=10311. To: sip:00359123456@12.34.56.78. Call-ID: 20495(a)34.56.87.12. Contact: sip:6789@34.56.87.12:5060. CSeq: 8112 INVITE. Max-Forwards: 70. Content-Length: 119. User-Agent: DUV1000 (dlk2.62)6789. Content-Type: application/sdp. . v=0. o=dlink 4526 9013 IN IP4 34.56.87.12. s=Session SDP. c=IN IP4 34.56.87.12. t=0 0. m=audio 6868 RTP/AVP 4 4 4. . # U 12.34.56.78:5060 -> 34.56.87.12:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 34.56.87.12:5060. From: 6789 <sip:6789@12.34.56.78>;tag=10311. To: sip:00359123456@12.34.56.78;tag=89fc68a3caf42d7e2d6038f5511e0f83.d0d2. Call-ID: 20495(a)34.56.87.12. CSeq: 8112 INVITE. WWW-Authenticate: Digest realm="12.34.56.78", nonce="423ab89f16ace6d2361b2f280fbcc0a1e55f06e1", qop="auth". Server: Sip EXpress router (0.9.0 (i386/linux)). Content-Length: 0. Warning: 392 12.34.56.78:5060 "Noisy feedback tells: pid=29694 req_src_ip=34.56.87.12 req_src_port=5060 in_uri=sip:00359123456@12.34.56.78 out_uri=sip:00359123456@12.34.56.78 via_cnt==1". . # U 34.56.87.12:5060 -> 12.34.56.78:5060 ACK sip:00359123456@12.34.56.78 SIP/2.0. Via: SIP/2.0/UDP 34.56.87.12:5060. From: 6789 <sip:6789@12.34.56.78>;tag=10311. To: sip:00359123456@12.34.56.78. Call-ID: 20495(a)34.56.87.12. Contact: sip:6789@34.56.87.12:5060. CSeq: 8112 ACK. Max-Forwards: 70. Content-Length: 0. User-Agent: DUV1000 (dlk2.62)6789. . # U 34.56.87.12:5060 -> 12.34.56.78:5060 INVITE sip:00359123456@12.34.56.78 SIP/2.0. Via: SIP/2.0/UDP 34.56.87.12:5060. From: 6789 <sip:6789@12.34.56.78>;tag=17031. To: sip:00359123456@12.34.56.78. Call-ID: 30054(a)34.56.87.12. Contact: sip:6789@34.56.87.12:5060. CSeq: 8288 INVITE. Max-Forwards: 70. Content-Length: 119. Authorization: Digest username="6789",realm="12.34.56.78",nonce="423ab89f16ace6d2361b2f280fbcc0a1e55f06e1",uri="sip:00359123456@12.34.56.78",algorithm=MD5,qop="auth",nc=00000006,cnonce="11367",response="ef6972dc5229d88060a952c10bdb2789". User-Agent: DUV1000 (dlk2.62)6789. Content-Type: application/sdp. . v=0. o=dlink 4526 9013 IN IP4 34.56.87.12. s=Session SDP. c=IN IP4 34.56.87.12. t=0 0. m=audio 7044 RTP/AVP 4 4 4. It seems that DUV sends authorization information ... but not correct maybe. Without www_autorize and proxy_authorize the device is able to place calls. Thanks in advance, Pavel

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[Serusers] Kphone, TLS and SER ... bug, and patch
by Cesc Santasusana 18 Mar '05

18 Mar '05

Hi, With the advent of TLS support for SER, i was testing it with KPhone (TLS being used between SER proxies only, Kphone-to-SER via UDP). Turned out that kphone would not respond to any message (it would receive them, but plain ignore them). The problem: TLS as transport mode in the VIA headers was not supported, thus, the SIP messages were considered not valid. I am attaching a patch to correct this for kphone 4.1.0, but i guess it can be applied with little modifications to earlier versions (4.0.3 to 4.0.5) but i am not sure. Regards, Cesc PATCH STARTS HERE: Index: dissipate2/sipclient.cpp===================================================================--- dissipate2/sipclient.cpp (revision 1)+++ dissipate2/sipclient.cpp (working copy)@@ -1034,6 +1034,9 @@ printf( "SipClient: Sending TCP Response\n" ); outsocket = new TCPMessageSocket; break;+ case SipVia::TLS:+ printf( "SipClient: TLS in top via, not supported (full TLS support not implemented)\n" );+ break; case SipVia::BadTransport: printf( "SipClient: Bad transport on incoming Via\n" ); break;@@ -1232,6 +1235,9 @@ printf( "SipClient: Sending TCP Response\n" ); outsocket = new TCPMessageSocket; break;+ case SipVia::TLS:+ printf( "SipClient: TLS in top via, not supported (full TLS support not implemented)\n" );+ break; case SipVia::BadTransport: printf( "SipClient: Bad transport on incoming Via\n" ); break;Index: dissipate2/sipvia.h===================================================================--- dissipate2/sipvia.h (revision 1)+++ dissipate2/sipvia.h (working copy)@@ -49,6 +49,7 @@ enum Transport { UDP, TCP,+ TLS, BadTransport }; /**Index: dissipate2/sipvia.cpp===================================================================--- dissipate2/sipvia.cpp (revision 1)+++ dissipate2/sipvia.cpp (working copy)@@ -37,6 +37,7 @@ switch ( t ) { case UDP: return "UDP"; case TCP: return "TCP";+ case TLS: return "TLS"; case BadTransport: return "BAD"; } return QString::null;@@ -46,6 +47,7 @@ { if ( t.compare( getTransportString( UDP ) ) == 0 ) { return UDP; } if ( t.compare( getTransportString( TCP ) ) == 0 ) { return TCP; };+ if ( t.compare( getTransportString( TLS ) ) == 0 ) { return TLS; }; return BadTransport; } PATCH ENDS HERE Unclassified

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[Serusers] How to forward calls to Cisco AS5300 GW
by Steven Xu 18 Mar '05

18 Mar '05

Hi serusers, Anybody has a sample ser.cfg that forward calls to Cisco AS5300 GW? I am using the Nathelper templet config file and got NAT +RTP relay works on IP2IP, but dont know how to work with PSTN through Cisco AS5300 GW. where to put the forwording code. I got some sample code here: # is it a PSTN destination? (is username nummerical and does not begin with 8?) if (uri=~"^sip:[0-79][0-9]*@") { # ... forward to gateways then; # check first to which PSTN destination the requests goes; # if it is US (prefix "1"), use the gateway XXX.XXX.XXX.XXX... if (uri=~"^sip:1") { # strip the leading "1" strip(1); forward(XXX.XXX.XXX.XXX, 5060); } else { # ... use the gateway YYY.YYY.YYY.YYY for all other destinations forward(YYY.YYY.YYY.YYY , 5060); } break; } I dont know if it can talk with Cisco 5300 GW though. Where should I put it in ser.cfg with NAThelper + RTPreply support? How to debug the ser.cfg, if i hear busy tone when i dial the PSTN number. Thanks for your time, Steve --------------------------------- Do you Yahoo!? Yahoo! Small Business - Try our new resources site!

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Re: [Serusers] Intermittent Audio
by pat newham 18 Mar '05

18 Mar '05

Hi Greger, The system is currently being tested by someone else but I believe they are behind a Linksys VPN router. Are you suggesting it could simply be the settings in this? I "think" I understand the nat issues associated with sip and sdp fairly ok so would I be correct in saying that if my two clients are behind nat(the same nat)on the same subnet the rtpproxy should be invoked? This would be my understanding of the situation but then I saw a recent email (see message header below)which suggests an external script should be used. Re: FW: [Serusers] calls between UA´s b ehind same NAT us ing nathelper/rtpproxy Also what confuses me is that the scenario works sometimes and yet other times it doesnt. I will attempt to get a full message dump (of both the working and non working scenario)from the tester if that will help. Kindest Regards, Pat. --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > Pat, > You haven't said anything of the type of NAT you are > behind. To me it sounds > like an ALG (Application layer gateway) problem. Try > to turn of the SIP ALG > in your router. If not, please post a full SIP > message exchange. You need > to find out if they communicate through the NAT > (hairpin media) or directly. > That depends on the SDP payload in the INVITE and OK > messages. > The new Getting Started document on > http://onsip.org/ (you need to > register) has a thorough review of NAT issues and > rewriting. Recommend! (I > wrote it ;-) ) > g-) > > pat newham wrote: > > Following on from my below email, I can now > definately > > say the problem is not nat pings. Just to recap I > am > > experiencing intermittent audio. It works when the > > phones have very recently registered, then > sometimes > > theres one way audio and then sometimes no audio. > Does > > anyone have any ideas what the problem could be or > > where I could begin to troubleshoot this? > > > > Hi, > > > > I have a strange problem. I have two grandstream > > budgetone clients on the same subnet behind nat > > registering with ser on a public address. > Obviously > > their public addresses would be the same but they > > listen on different ports. When they initially > > register, I can the call,audio is transmitted and > > everything is successful. > > > > However sometimes theres only one way audio, other > > times theres no audio and then other times it > > works....I am guessing that this is because the > nat > > router is forgetting the nat mapping so after a > while > > when the nat mapping is "forgotten" and a packet > > arrives destined for a client, the router drops > it.... > > > > Could someone verify this for me??...Am I on the > right > > track?? I have the following settings in ser.cfg > which > > I thought would keep the nat settings alive. > > > > 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 > > > > I also increased the nat keep alives "pings" sent > in > > the configuration settings of the grandstream > > phone....Any further ideas?? > > > > Regards, > > Pat. > > > > Send instant messages to your online friends > > http://uk.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > Send instant messages to your online friends http://uk.messenger.yahoo.com

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[Serusers] about a project (SER-CME)
by Andrea Riela 18 Mar '05

18 Mar '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I'm really newbie on SER, but I've a project in my mind, and I would your advices and suggestions, if you please. Situation: I've multiple SIP accounts, and my Cisco CME could register only one with sip-ua and a dial-peer voip. And the Cisco CME hasn't itself a voicemail solution. So, my idea is: register the SIP accounts on the SER, then forward all calls to an internal SIP number that I'll register to CME. Then, if I need a voicemail, I could forward calls from CME to SER (there's a voicemail feature? could I try to record the call and play back with my ip phone and/or send the wav file as attach via email?). I hope you could put me in the right direction. Thanks for any support, question or advice. Regards Andrea -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCOqyPMakHrsrHP9wRArkaAKCWryEIT18EE6Hw0oZWfMfqEe7ZyACfZYG7 CqeN0sTG87QSPo/rK0cvXfE= =+9eJ -----END PGP SIGNATURE-----

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[Serusers] State field in the location table
by Jffdsaklj dsjk 18 Mar '05

18 Mar '05

What's the meaning of the state field in the location table?. Usually the value is 0. But sometimes, its value is 1 and then we have some problems in our applications. What's the difference?. Thanks

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[Serusers] enum - e164.org
by Pamela Weis 18 Mar '05

18 Mar '05

hello group, has anyone experience with ser's enum module and tried it with e164.org? i've something like in my ser.cfg: if(enum_query("e164.org")){ .... }; i tried to dial a number which is known there but I won't get into this if clause. but when i try if(enum_query("e164.arpa")){ .... }; and call a number known there the enum_query works? any help with enum would be greatly appreciated. thanks in advance. pamela

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[Serusers] Hold/Unhold doesn't work with NATed clients with rtpproxy in some case...
by jimmy 18 Mar '05

18 Mar '05

Hi: I had a very basic doubt with respect to the function "hold/unhold" on ser with rtpptoxy and kphone. Do nathelper and rtpproxy currently support hold/unhold function (with kphone 4.1.0) with nated clients ? Or do they have other methods to acheive that function? In my test, I put rtpproxy and ser in same pc, and two UACs are behind that pc, When I use hold/unhold, most time it works but some times it doesn't work I checked the ethereal log, it seems that the fail happens only when 1. hold function last less than 60 seconds. 2. once the re-invite's rtp port in SDP announced by client has been changed to different rtp port (different from original rtp port). So it seems rtpproxy doens't knows the ports come from both UACs have changed... and didn't forward UACs' rtp packets. Here is the question I have conclude from above description 1. While receive the re-invite message, will ser recheck the port in SDP and announce rtpproxy again? 2. Is this the limitation of Ser+rtpproxy with NATed UACs? 3. Does there has any suggestion that can make this function works? (ser+nathelper+rtpproxy with function hold/uhhold) Thanks and best regards Jimmy

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[Serusers] serweb login problem
by Carlos Vicente 18 Mar '05

18 Mar '05

Many people have reported having problems logging in to serweb after installation. I had the problem myself, and after some inspection of the html/admin/index.php file, the following lines game me a hint: if ($config->clear_text_pw) { $q="select phplib_id from ". $config->table_subscriber. " where username='".addslashes($uname)."' and password='".addslashes($passw)."' and perms='admin' and domain='".addslashes($config->realm)."'"; So, the SELECT statament wants to match the 'username', 'password', 'perms' and 'domain' fields from the 'subscriber' table. In your html/config.php file, comment-out the regex replace and type your domain exactly as it is in your db. e.g: /* your domain name */ # $this->realm=$this->domainname=$this->default_domain= ereg_replace( "(www\.|sip\.)?(.*)", "\\2", $_SERVER['SERVER_NAME']); $this->realm="mydomain.com"; Hope this helps. cv

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[Serusers] request queue
by kcassidy＠kakelma.mine.nu 18 Mar '05

18 Mar '05

Hi All, I have a strange problem with my ser-0.8.14. I noticed that sometimes it's very slow in responding to requests (e.g. INVITES). After tinkering around, I saw my UDP queue gets filled up (netstat -an). The only fix I have right now is to restart the process. Then the problem will come up again around 1-2 days. I have around 40 registered users, and tried setting the children parameter in ser.cfg to different numbers like 4,6, and 10. But still having the problem. Any ideas? Thanks.

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[Serusers] 0.9.0 rpid bug?
by Daniel Poulsen 18 Mar '05

18 Mar '05

Perhaps this is a bug, perhaps it is by design? In ser 0.9.0, when i set debug=3, rpid ceases to function. Ethereal shows: [debug=3] Remote-Party-ID: 66570ab07228: authentication acc;party=calling;id-type=subscriber;privacy=none;screen=no [debug=2] Remote-Party-ID: sip:2125551212@sip.myhost.net;party=calling;id-type=subscriber;privacy=off;screen=no If this is a bug, let me know if I can provide more info. Dan

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[Serusers] rfc3261 and ACKs
by Steve Blair 17 Mar '05

17 Mar '05

Hello: Me again. I'm still stuck in the maze of call transfer and call forward all problems. Our gateway vendor suggested a new code release. Using this release completely breaks "normal" 5-digit and other calls that traverse the gateway. The vendor may be onto the problem but their assessment points back to my SER proxy. Given that the proxy works fine in all other calling scenarios with the old gateway code I'd like to know what is significant about their new code and SER. By the way the proxy in question is running 0.8.14 stable. The vendor problem description follows: Thanks,Steve --- cut here --- Here is what I believe the summary of the problem. I believe that the SIP Proxy is not complying with rfc3261 on how to process the ACK when Loose Routing is enabled. Debug Snippet from 12.3(13) ======================= *Mar 2 00:02:16.546: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 128.91.56.38:5060 From: <sip:343408@128.91.56.38>;tag=5286C18-13B5 To: <sip:68001@net.isc.upenn.edu>;tag=003094c38fd4003a29fa1f0f-264a55c1 Call-ID: 2C5BA6F-15B811CC-8122ED93-3017C386(a)128.91.56.38 Date: Thu, 17 Mar 2005 12:08:16 GMT CSeq: 101 INVITE Server: CSCO/7 Contact: <sip:68001@128.91.56.6:5060> Record-Route: <sip:68001@128.91.56.47;ftag=5286C18-13B5;lr=on> *Mar 2 00:02:16.578: Sent: ACK sip:68001@128.91.56.47:5060;ftag=5286C18-13B5;lr=on SIP/2.0 Via: SIP/2.0/UDP 128.91.56.38:5060 From: <sip:343408@128.91.56.38>;tag=5286C18-13B5 To: <sip:68001@net.isc.upenn.edu>;tag=003094c38fd4003a29fa1f0f-264a55c1 Date: Tue, 02 Mar 1993 00:02:15 GMT Call-ID: 2C5BA6F-15B811CC-8122ED93-3017C386(a)128.91.56.38 Route: <sip:68001@128.91.56.6:5060> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ The IP Phones address and that is why the call proceeds further. Because the ACK gets to the phone Debug Snippet from 12.3(11)T3 ========================= As of this code release, the gateways added RFC3261 support for "Loose Routing". Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 128.91.56.38:5060;branch=z9hG4bK0F86 From: <sip:4843433408@128.91.56.38>;tag=60040-250 To: <sip:68001@net.isc.upenn.edu>;tag=003094c38fd4003c61e999c2-43d890a2 Call-ID: 4479EFAB-2BF611D6-800BA01F-9B2E2BA3(a)128.91.56.38 Date: Thu, 17 Mar 2005 12:30:36 GMT CSeq: 101 INVITE Server: CSCO/7 Contact: <sip:68001@128.91.56.6:5060> Record-Route: <sip:68001@128.91.56.47;ftag=60040-250;lr=on> *Mar 1 02:52:58.151: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:68001@128.91.56.6:5060 SIP/2.0 Via: SIP/2.0/UDP 128.91.56.38:5060;branch=z9hG4bK116B8 From: <sip:4843433408@128.91.56.38>;tag=60040-250 To: <sip:68001@net.isc.upenn.edu>;tag=003094c38fd4003c61e999c2-43d890a2 Date: Fri, 01 Mar 2002 02:52:57 GMT Call-ID: 4479EFAB-2BF611D6-800BA01F-9B2E2BA3(a)128.91.56.38 Route: <sip:68001@128.91.56.47;ftag=60040-250;lr=on> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Address of the Loose Router which is compliant to route processing in RFC3261 The SIP Proxy (Which is the loose router) should receive this ACK and forward onto the IP Phone or the address in the Req-URI which it isn't. The proxy is actually acting like a Strict router in this instance. The Proxy is adding the loose routing parameters in the headers but doesn't seem to processing the Requests correctly as a loose router. So that is why we are having issues with a basic call in this code release.

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[Serusers] I need serial forking example
by PHAN, Quang-Minh (Docteur Ordinateur) 17 Mar '05

17 Mar '05

Hi List, Can anyone send me an example of serial forking using avpops module. I remember reading one tutorial somewhere but I couldn't find it again. Thanks for your help Minh

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[Serusers] Problems inspecting SER with ngrep
by Alberto Cruz 17 Mar '05

17 Mar '05

Hi all, I'm trying to inspect the packets that are arriving to my SER server with ngrep but when I executed the command I got the following: [root@matrix root]# ngrep @sip.telereunion.com.mx port 5060 interface: eth0 (65.208.39.208/255.255.255.240) filter: ip and ( port 5060 ) match: @sip.telereunion.com.mx ## As you can see I'm only getting hash symbol. Does anybody know how could I display the right info? I'm running RedHat Linux 9.0, ser 0.9.1, ngrep 1.43 and libcap 0.8.3 Regards Alberto Cruz

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[Serusers] additional contact info from grandstream budgetone ?
by maka 17 Mar '05

17 Mar '05

Hi all, i just got a gradnstream budgetone for testing, and I see some strange behaviour on its part - it 's adding an additional ;user=phone string to the contact field it sends to the registrar, and it thus produces some bad header errors when communicating with the phone. the phone also gets registered twice for the same number -- once with the extra information from above, and once without it. has anyone dealt with something like this?

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[Serusers] Syslog options
by Marc Khayat 17 Mar '05

17 Mar '05

hi all, i'm using ser-0.8.14, and i'm receiving call logs on a syslog server. But i'm not getting the call duration: in the below logs, SER is linux.domain.net, 192.168.0.150 is registered, 65.25.48.112 is not registered but stateful routing is used (t_relay_to_udp) Mar 17 17:22:56 linux /sbin/ser[10908]: ACC: transaction answered: method=INVITE, i-uri=sip:33@linux.domain.net, o-uri=sip:33@linux.domain.net, call_id=0318F319-981E-4AD7-86E8-1DA55DB661C3(a)192.168.0.150, from=Marc Khayat <sip:777@linux.domain.net:5061>;tag=3813796527, code=200, to=<sip:33@linux.domain.net>;tag=b5422284a4 Mar 17 17:22:56 linux /sbin/ser[10912]: ACC: request acknowledged: method=ACK, i-uri=sip:33@65.25.48.112, o-uri=sip:33@65.25.48.112, call_id=0318F319-981E-4AD7-86E8-1DA55DB661C3(a)192.168.0.150, from=Marc Khayat <sip:777@linux.domain.net:5061>;tag=3813796527, code=200, to=<sip:33@linux.domain.net>;tag=b5422284a4 Mar 17 17:22:58 linux /sbin/ser[10915]: ACC: transaction answered: method=BYE, i-uri=sip:33@65.25.48.112, o-uri=sip:33@65.25.48.112, call_id=0318F319-981E-4AD7-86E8-1DA55DB661C3(a)192.168.0.150, from=Marc Khayat <sip:777@linux.domain.net:5061>;tag=3813796527, code=200, to=<sip:33@linux.domain.net>;tag=b5422284a4 I tried the options in log_fmt, but there's nothing. How is it best to do accounting if syslog will not give me the call duration? would there be any difference (call duration, ASR, ...) if I use a Radius or MySQL ? Thanks!

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RE: [Serusers] Mediaproxy consuming RAM resources.
by Ricardo Martinez 17 Mar '05

17 Mar '05

Paul. Hello. Sorry to interrupt you. I'm still stuck with the mediaproxy resource problem. I've installed the patch from the pyhton.org but the problem seems to persist. I'm thinking maybe the problem is related with the way i treat a NAT'd call. I hope that you can help me with some questions. I have a mix NAT'd solution, the mayority of my endpoint have STUN enabled, so if this first solution fails, the mediaproxy is enabled for those calls. Regarding to the mediaproxy : What i do is to check if the caller is NAT'd if so, the mediaproxy is enabled. When a endpoint with a public ip call to a NAT'd endpoint i do the same check but in the on_reply_route. Is this ok?. How you handle a call when a endpoint in the public internet call to another in the public internet too? Do you use mediaproxy in that case?. I'm attaching my ser.cfg to check if there is any problem with it. I really appreciate any help Regards, Ricardo.- > -----Mensaje original----- > De: Java Rockx [mailto:javarockx@gmail.com] > Enviado el: Miércoles, 16 de Marzo de 2005 10:22 > Para: Ricardo Martinez > CC: serusers(a)lists.iptel.org > Asunto: Re: [Serusers] Mediaproxy consuming RAM resources. > > > That's what I have. I'm not sure what the setflag(1) is for, but then > again, I cannot see the entire ser.cfg. > > Regards, > Paul > > > On Wed, 16 Mar 2005 10:20:41 -0400, Ricardo Martinez > <rmartinez(a)redvoiss.net> wrote: > > Hello Paul. > > I have the next lines for ending media session: > > > > if (loose_route()) { > > if (method=="BYE" || method=="CANCEL") { > > end_media_session(); > > setflag(1); > > }; > > t_relay(); > > break; > > }; > > > > Is this ok?. I got the idea from a ser.cfg file in the > forum. I'm not so > > sure if this is correct. > > Can someone help me here? > > > > Thanks again! > > > > Regards, > > Ricardo > > > > > -----Mensaje original----- > > > De: Java Rockx [mailto:javarockx@gmail.com] > > > Enviado el: Martes, 15 de Marzo de 2005 21:03 > > > Para: serusers(a)lists.iptel.org > > > Asunto: Re: [Serusers] Mediaproxy consuming RAM resources. > > > > > > > > > Yes that could help, but if you're experiencing pleny of > "idle" RTP > > > connections then you are probably not ending your media > sessions with > > > end_media_session(). > > > > > > Regards, > > > Paul > > > > > > > > > On Tue, 15 Mar 2005 18:33:19 -0600, Alberto Cruz > > > <acruz(a)tekbrain.com> wrote: > > > > If you are running RedHat 9 and Python 3.2.5 may be this > > > could help you: > > > > > > > > http://www.python.org/download/download_linux.html > > > > > > > > Regards > > > > > > > > Alberto Cruz > > > > > > > > > > > > Ricardo Martinez wrote: > > > > > > > > Hello. Returning to this subject. I upgraded to the new > > > mediaproxy version. > > > > 1.2.1, but i still have the same problem. I detected a > > > increasing amount of > > > > RAM memory used by mediaproxy. I don't know what's > > > happening. Is this > > > > something maybe related to the way i tear dow the session, > > > i'm still confuse > > > > with so many IDLE session in my mediaproxy. Can this be the > > > cause of the > > > > increasing amount of RAM memory used? Also, is there a way > > > to debug the > > > > mediaproxy? or something that help me to understand what's > > > going on? Any > > > > help would be appreciated. Thanks Ricardo.- > > > > -----Mensaje original----- De: Java Rockx > > > [mailto:javarockx@gmail.com] > > > > Enviado el: Viernes, 11 de Marzo de 2005 15:23 Para: > > > Ricardo Martinez CC: > > > > serusers(a)lists.iptel.org Asunto: Re: [Serusers] Mediaproxy > consuming RAM > > > > resources. I'm tearing down my sessions the same way you > > > showed in your > > > > first email. So AFAIK, that is OK. Regards, Paul On > Fri, 11 Mar 2005 > > > > 16:15:38 -0400, Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > > > > Paul, I was thinking the same, maybe that's the problem. > > > What about the > > > > tear down of the sessions? Thanks! Regards, Ricardo.- > > > > -----Mensaje original----- De: Java Rockx > > > [mailto:javarockx@gmail.com] > > > > Enviado el: Viernes, 11 de Marzo de 2005 15:09 Para: > > > Ricardo Martinez CC: > > > > serusers(a)lists.iptel.org Asunto: Re: [Serusers] Mediaproxy > consuming RAM > > > > resources. Ricardo, I think the release notes on > > > mediaproxy-1.2 say that > > > > some resource leaks have been fixed. Perhaps you can > try version 1.2 > > > > Regards, Paul On Fri, 11 Mar 2005 16:01:59 -0400, > Ricardo Martinez > > > > <rmartinez(a)redvoiss.net> wrote: > > > > Hello list. I was looking the process in my SER > sip-proxy with the > > > > mediaproxy running. > > > > Every time i look, the percetage of used RAM is > > > increasing. It seems like > > > > for some reason mediaproxy is not releasing the resources > > > used in RAM. You > > > > can look this in the next lines : PID USER PRI NI SIZE RSS > > > SHARE STAT %CPU > > > > %MEM TIME CPU COMMAND > > > > 1952 root 15 0 284M 266M 1628 S 0.0 26.4 19:22 0 > > > > mediaproxy.py Can someone tell me if this is a bug of > > > mediaproxy or maybe a > > > > missconfiguration in my ser.cfg? I want to ask a question > > > regarding to tear > > > > down sessions with mediaproxy > > > > also. In my ser.cfg i have : if (loose_route()) { if > > > (method=="BYE" || > > > > method=="CANCEL") { log(1, "NAT: BYE o CANCEL recibido --> > > > terminando la > > > > sesion de media\n"); end_media_session(); setflag(1); }; > > > t_relay(); break; > > > > }; Is this ok? I think in some cases (i'm still unable to > > > determine which > > > > ones), SER does not tear down a mediaproxy session. You > > > can see for example > > > > this two session, both of them were ended, but it seems to > > > be active even > > > > before that. (i ommited the source, via and destination > > > IP's in the output > > > > of the session.py) Status Duration Codec Type Traffic > > > > -------------------------------------------------------------- > > > > -------------- > > > > --- hold (2430) 40'40" G729 Audio 11.53k/29.58k/18.45k > > > inactive 0'02" > > > > Unknown Audio 0/0/0 Can someone tell me what's happening? > > > I'm runnig : [root > > > > mediaproxy]# ./mediaproxy.py --version mediaproxy.py 1.0 > > > Thanks in advance. > > > > Regards, Ricardo M. > > > _______________________________________________ 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 > > > > _______________________________________________ > > > > 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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[Serusers] just noticed a strange problem with aliases...they dont work instantly
by Iqbal 17 Mar '05

17 Mar '05

Hi If I add a alias manually into my DB, they dont seem to go live instantly, serctl, doesnt show em...I guess thats because its reading from memory, if this is the case, then how can I get the entries in the DB into memory...without restarting. Or shoudk I not add directly into the DB, and go via serctl...which is not really want I want to do... Iqbal

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RE: [Serusers] LCR gw config problem?
by Matt Schulte 17 Mar '05

17 Mar '05

smallint doesn't seem to allow null, when I change to varchar to allow null it breaks the whole deal... -----Original Message----- From: Juha Heinanen [mailto:jh@lohi.tutpro.com] Sent: Thursday, March 17, 2005 8:47 AM To: Matt Schulte Cc: serusers(a)lists.iptel.org Subject: [Serusers] LCR gw config problem? Matt Schulte writes: > I'll keep it short, when trying to use the gw lookup everything works > fine except the port gets altered to some crazy random port number. > Thoughts? hard to say. it works for me. if you use port 5060, you can give port field NULL value. -- juha

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[Serusers] consume_credentials() not removing Proxy-Authorization Header
by Java Rockx 17 Mar '05

17 Mar '05

Hi All. Should consume_credentials() remove the Proxy-Authorization header from my SIP messages before I t_relay() them? If not, how should I remove the header? Use remove_hf() ? Regards, Paul

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Re: [Serusers] Install from sources howto
by lakmal silva 17 Mar '05

17 Mar '05

Hi Oliver, take a look at these links. It will give you a quick intro to SER intallation and configuration. http://www.iptel.org/ser/doc/ser-howto/ser-Howto.html http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/INSTALL… To untar the tarball, you can issue the following command. #tar -xzvf ser-0.xx_src.tar.gz Alternatively you can download the source code from the cvs tree. Regards, Lakmal --- Olivier Taylor <olivier.taylor(a)gmail.com> wrote: > Hi all, > > If I understand well, to use all the features of > Ser, it's necessary to > build Ser from sources and install it. > I used rpms to install ser, I did configure it, > everythings ok, but now I > wanna do more now. > > Is there somewhere an installation guide from > source, how to untar, to > compile and so on? > > Thx, > > Olivier > > -----Message d'origine----- > De : serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] De la > part de Aldi Ramdhani > Envoyé : dimanche 13 mars 2005 12:36 > À : serusers(a)lists.iptel.org > Objet : [Serusers] window messenger 5.0 > > > Dear All, > > I'm running ser-0.8.14 on FreeBSD 5.3-RELEASE. I > install it through ports > collection with MySQL support. I try ro login using > Window Messenger 5.0. It > works, but i see other user were offline even they > are online. can somebody > tell me what's wrong ??? > > _______________________________________________ > 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 > __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

3 3

[Serusers] LCR gw config problem?
by Matt Schulte 17 Mar '05

17 Mar '05

I'll keep it short, when trying to use the gw lookup everything works fine except the port gets altered to some crazy random port number. Thoughts? (after load_gw..) Time Thu Mar 17 08:12:59 2005 RURI <sip:+13143212222@209.247.17.5:166486261> +--------------------+----------+------+--------+ | gw_name | ip_addr | port | grp_id | +--------------------+----------+------+--------+ | blah | 85063633 | 5060 | 1 | +--------------------+----------+------+--------+ port is set to smallint(unsigned) .. wtf am I doing wrong? :-) Ty..

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[Serusers] failure with rtpproxy: send_rtpp_command: can't read reply from a RTP proxy
by Michael Storz 17 Mar '05

17 Mar '05

Hello all, I have integrated the nathelper-module in SER. This I have done such as on this site: http://www.terena.nl/library/IPTELEPHONYCOOKBOOK/contents1.html, chapter 4.7.3.5 I have installed this rtpproxy: http://www.portaone.com/~sobomax/rtpproxy.tar <http://www.portaone.com/%7Esobomax/rtpproxy.tar> and started it. Now it is always this failure, when I restart SER: 3(9842) ERROR: send_rtpp_command: can't read reply from a RTP proxy 3(9842) WARNING: rtpp_test: can't get version of the RTP proxy 3(9842) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily How can I solve this problem? Can somebody help me? Michael

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