- sr-users - kamailio.org

[Serusers] auth.so modules and digest auth

by Benny Ho

I load the auth.so and the default config. Uncomment those to use auth.so But when I call up two user agents not register in the database. The only difference is there a warning for registration. They are still calling each other. Why?

21 years, 4 months

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[Serusers] postgres db support

by lgfausak＠august.net

Has anybody started on support for Postgres? I've checked out the layering and it seems to be a fairly modular project. I haven't coded in C for quite some time, but I think I can handle it. Any hints? Should I even try? The only thing I can see having problems with is the build/configure system. How do I hook in a new module? How are include directories and new libraries added? I was pretty good with make, gmake, imake in my day but I can't seem to find the head of the Make tree! ---greg Greg Fausak

21 years, 4 months

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[Serusers] what makes a transaction?

by lgfausak＠august.net

I'm trying to understand stateful vs. stateless transactions. A stateless 'transaction' simply lasts as long as it takes to process the current sip request. It is then forgotten. However, a stateful transaction I'm having a little problem grasping. For instance, if I t_relay() an INVITE, logically the transaction is started....so when the response comes it isn't delivered to the main route {} block, but instead it is delivered to the awaiting stateful t_relay(), probably keyed by callid. So, for instance: INVITE <- 401 Unauthorized -> Is this entire sequence a transaction? Right after this the UA sends an ACK, this is a transaction unto itself? ACK <- The UA then sends : INVITE <- 100 Trying -> 180 Ringing -> 200 OK -> All of these messages are a 'transaction' handled by t_relay()? The next transaction is: ACK <- Finally, the last transaction would be (as an example): BYE <- 200 OK -> So the ser route{} block would need to recognize and handle INVITE ACK BYE in this context. Is that correct? If I was running 'statelessly' I would need to handle and send every single message listed? ---greg

21 years, 4 months

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[Serusers] Segmentation fault: fifo_server fgets failed: Illegal seek

by jaime.gill＠orange.co.uk

Hi,

21 years, 4 months

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[Serusers] ACC's ACK

by lgfausak＠august.net

I realize the accounting logging of ACK is only an attempt, but ... It does seem fairly important in determining when the actual call started. Here is some filtered output from the syslog accounting output file: --- -------- -------- -------- --- ---------- ---------- Call INVITE ACC BYE Length From To --- -------- -------- -------- --- ---------- ---------- 1 18:50:39 18:50:39 18:50:48 9 4695461243 -> 9723236598 2 18:53:19 18:53:19 19:08:46 927 4695461243 -> 9723937976 3 20:14:00 0 4695461246 -> 4695461235 4 20:16:41 20:16:48 7 4695461246 -> 4695461235 5 20:19:09 20:19:17 8 4695461246 -> 4695461235 6 20:30:18 20:30:23 5 4695461235 -> 4695461235 7 20:30:51 20:31:01 10 4695461246 -> 4695461235 8 20:37:14 20:37:17 3 4695461246 -> 4695461235 9 21:02:08 21:03:28 80 4695461235 -> 4695461240 10 21:21:26 21:21:26 21:36:33 907 4695461235 -> 4695461243 11 22:42:03 22:42:03 22:52:23 620 4695461235 -> 4695461235 12 22:59:22 22:59:33 11 4695461246 -> 4695461235 13 23:00:46 23:01:00 14 4695461246 -> 4695461235 14 23:08:11 23:08:21 10 4695461246 -> 4695461235 15 23:19:37 23:19:37 23:19:41 4 4695461235 -> 4695461235 16 23:22:49 23:22:56 7 4695461235 -> 4695461235 17 23:28:55 23:28:55 23:29:07 12 4695461235 -> 4695461243 18 23:29:18 23:29:18 23:42:53 815 4695461235 -> 4695461243 19 09:08:11 09:08:11 09:08:52 41 9725678129 -> 4695461236 20 09:09:50 09:10:38 48 4695461236 -> 99727139968 21 09:11:15 09:12:00 45 4695461236 -> 99727139968 22 09:12:35 09:12:35 09:13:39 64 2143357976 -> 4695461240 23 09:18:57 09:18:57 09:19:29 32 4695461235 -> 9724620762 24 09:23:17 09:24:25 68 4695461235 -> 4695461240 --- -------- -------- -------- --- ---------- ---------- I note the ACK times are all the same as the INVITE times. Isn't the wrong ACK being recorded? Is this a problem with my .cfg? A much more useful ACK would be the second one (when the far end ACKs after the 180 Ringing and 200 OK messages. A typical call, from UA perspective: INVITE -> 401 unauth.. <- ACK -> (evidently, the one being accounted) INVITE (with cred) -> 100 Trying <- 180 Ringing <- 200 OK <- ACK -> (the ACK that SHOULD be accounted, IMHO) ... ---greg

21 years, 4 months

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Re: [Serusers] no mysql authentication on solaris8

by Jiri Kuthan

Thanks a lot. It seems that the log line > 5(391) authorize(): Invalid nonce value received, very suspicious ! is the key -- Jan will probably handle that better than myself. Are you sure you applied the solaris/digest patch? (See nonce validation in ISSUES on SER's website. See also the topmost solaris-specific issue -- better take SER sources and recompile them without DBG_QM_MALLOC in Makefile.defs) If things don't work then, send me please captured SIP messages too. (Use tools like "ngrep port 5060" to record them.) Thanks, -Jiri

21 years, 4 months

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[Serusers] OT: sip ua question

by Tomas Björklund

Hi, A bit offtopic perhaps, but are there any good win32 sip ua's out there? I'm interested in both commercial and opensource software (my friends doesn't seem to whant to buy Cisco 7960's ... wonder why :) / Tomas

21 years, 4 months

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[Serusers] Redundant forwarding

by Jamin W. Collins

This is probably a very simple question. I'm trying to create a very simplistic configuration that simply forwards the call to one of two fixed destinations. Essentially, it forwards all calls to server A and if that fails, forwards the call to server B. Forwarding to server A is no problem. Within route, I simply have: if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; rewriteFromRoute(); if (!forward("192.168.1.1")) { sl_reply_error(); }; This works perfectly, but only forwards to one server. I tried changing the last bit to: if (!forward("192.168.1.1")) { if (!forward("192.168.0.1")) { sl_reply_error(); }; }; However, this results in "ERROR: bad config file (5 errors)". Any suggestions? Is something like this possible? -- Jamin W. Collins

21 years, 4 months

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RE: [Serusers] problem in using acc.so and radius_acc.so and auth.so

by Benny Ho

Dear Jiri, I have successfully compile the acc.so with SQL support. And I run a number of testing on it. It seems works fine but I encounter the following error which crach the ser. =========================================================== [root@md benny]# 0(3545) mod_init(): Database connection opened successfuly Maxfwd module- initializing rr - initializing acc - initializing stateless - initializing mysql - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(3565) INFO: fifo process starting: 3565 9(3565) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 10(3574) ACC: transaction answered: method=INVITE, i-uri=sip:1000@md.com;user=ph one, o-uri=sip:1000@192.168.10.18:5000;user=phone, call_id= 426a69eafe94e7923666 1278d86d5024(a)192.168.10.18, from= UserAgent-1<sip:1000@192.168.10.18:5000;user=p hone>, code=408 10(3574) use_table(): Invalid parameter value 0(3545) child process 3574 exited by a signal 11 0(3545) core was not generated 0(3545) INFO: terminating due to SIGCHLD ========================================================================= It seems to be a crash on use_table() when code=408, I tried to unanswer to call or call to itself, and this problem generated. Benny -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Monday, February 24, 2003 6:30 PM To: Benny Ho Subject: RE: [Serusers] problem in using acc.so and radius_acc.so and auth.so First of all, note well: http://lists.iptel.org/pipermail/serusers/2003-February/000493.html At 04:36 AM 2/24/2003, Benny Ho wrote: >Could you tell me how to enable SQL_ACC def? and setting teh db_url >parameter? uncomment the "#DEFS+=-DSQL_ACC" line in modules/acc/Makefile >I cannot find any doc for mysql modules. mysql module is not used from ser scripts directly -- modules which rely on mysql use it internally. acc w/mysql has not been released yet, it is work in progress on CVS and it has not been documented. -jiri

21 years, 4 months

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Re: [Serusers] Redundant forwarding

by lgfausak＠august.net

I'm currently working with SNOM to get this issue fixed. I'll report when I know something. It might not hurt if someone else reported it to them! ---greg Greg Fausak > >At 09:34 PM 2/24/2003, Jamin W. Collins wrote: >>On Mon, Feb 24, 2003 at 08:54:00PM +0100, Jiri Kuthan wrote: >>> What is CIC? >> >>A PBX made by I3 (www.inin.com) >> >>> Send us network dumps please -- that will make it easier to understand >>> for us. You can for example use "ngrep port 5060". >> >>Network dump from the SER server is attached. > >snom phone is a common suspect again. > >The first signaling transaction looks ok. The only unusual thing is that >for some reason, the ININ server advertises 192.168.0.9 as media receiver >whereas its signaling lives at 192.168.1.16. Is media decomposed? > >The really suspicious is the second transaction. The snom phone sitting >at .36 advertises contacts with .34 in its replies. That makes the >caller (ININ) to send an ACK to .34. Snom does not received it and >keeps soliciting it by retransmitting 200s. > >-Jiri > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

21 years, 4 months

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[Serusers] Some thoughts and questions

by Tomas Björklund

Hi, I've been looking at ser for a couple of days now and there some things I can't find in the documentation. .1. Logging funktionality I'm used to being able to log more detail than what I can get out ser. I'm problably missing something in the documentation, but what I want is something like the following timestamp ser[pid] Incoming packet from ip:port [dump of packet (only the sip header/body)] timestamp ser[pid] Invite from [] to [] timestamp ser[pid] After rewrite: from [] to [] timestamp ser[pid] Relaying packet to [pstn-gw/other sip server] / timestamp ser[pid] Handling request for local user [] Can you send arguments to the log() function in the configuration file for example log(1,"Call from %s", uri:from); ?? .2. Rewrite rules Is it possible to rewrite the from uri? For example when I [sip:tomas@krixor.xy.org] want to call a pstn number I must rewrite the from address to [sip:myphonenumber@krixor.xy.org] before I send the request to the pstn gateway And when a incoming call comes from the pstn gateway I want to rewrite the from address from [<sip:number@ip-of-pstn-gateway>] to {"PSTN Luser" <sip:number@pstn-gw.mydomain.com>] I haven't looked at the source in the CVS yet, but I aim to do so soon, I saw in the list archive that you have lots of nice new features planned for the next release. ENUM support is one of the better ones :) Anyway, looking forward to trying to help with the development of this software. Best regards, Thomas Björklund

21 years, 4 months

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[Serusers] Rewriting URI in the Contact field

by Maxim Sobolev

Folks, I need an advise on how to better implement one feature, which isn't currently present in SER. We need to allow UAs behind NAT properly register with the registrar - by "properly" I mean that host:port portion of URI in Contact field should not be used, but host:port the request came from should be used instead. By definition we know that those UAs will support symmetric SIP signalling, so that this scheme will work just fine. In my opinion there are two ways to do it: either add new rewritecontact* family of functions similar to rewritehost ones. or add a new flag for the save() function. This is where I need your help - which implementation looks better for you (or maybe you have even some better idea), since we are really interested in inclusion of our changes into the mainline to reduce our local hacks. Regards, Maxim

21 years, 4 months

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[Serusers] Some patches for integration

by Maxim Sobolev

Folks, Attached please find several patches for ser 0.8.10. They do the following: 1. Removes hardcoded `/usr/local' from Makefiles and replaces it with LOCALBASE variable setable in the top Makefile.defs. This should make life of packagers easier. 2. Allows to override PREFIX variable with something else from environment. 3. Adds support for FreeBSD doc/man directories layout. 4. Adds some missed -I's into radius modules. 5. Fixes radius modules to use the same symbolic avp names as in dictionary. 6. Adds support for recording source port in msg structure and adds it in "Noisy feedback". This should be extremely useful for debugging various SIP&NAT scenarios and later could be re-used to implement `rport' support described in my previous message. It would be nice to have those patches integrated into the next release. Thanks! -Maxim

21 years, 4 months

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[Serusers] Record-route:/Contact: conflict?

by Greg Fausak

I've got a packet that looks like: # U 2003/02/24 07:56:52.503535 216.87.144.203:5060 -> 216.87.145.22:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-ng5tokyx448r. From: "snom man" <sip:4695461245@augustvoice.net>;tag=8u6ju8wxuc. To: <sip:2143357976@augustvoice.net;user=phone>;tag=3CBB0360-532. Date: Mon, 24 Feb 2003 13:56:43 GMT. Call-ID: 3c267202b6a8-lgseu8olovlp(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Session-Expires: 7200;refresher=uac. Require: timer. Allow-Events: telephone-event. Contact: <sip:92143357976@216.87.144.196:5060;user=phone>. Record-Route: <sip:2143357976@216.87.144.203;ftag=8u6ju8wxuc;lr>. Content-Type: application/sdp. Content-Length: 209. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7543 5694 IN IP4 216.87.144.196. s=SIP Call. c=IN IP4 216.87.144.196. t=0 0. m=audio 16632 RTP/AVP 0 100. a=rtpmap:0 PCMU/8000. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. The main thing to note is the packet is sent to a SNOM phone from my PROXY server. Curious, when the SNOM phone responds it is responding to 216.87.144.196, which is my GATEWAY. My GATEWAY is configured to only listen to the PROXY server, so it drops the packet. In the packet above, the Record-route line indicates that 216.87.144.203 should be responded to, which is my PROXY. Also in the packet above, there is a 'Contact:' line that indicates the GATEWAY at 216.87.144.196 should be contacted. Obviously the SNOM phone is getting the address from the Contact: line (or maybe the SDP) rather than the Record-route: line. Is it doing so incorrectly, or do I have a bug in my GATEWAY, or does the PROXY need to strip out the Contact: line?

21 years, 4 months

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[Serusers] SER parser benchmarks

by Sebastien Pierre

Hi! I was just wondering if you had some benchmark for the SER parser... as SER does not seem to offer a libser, it makes it more difficult to write benchmarking programs :/ TIA, -- Sebastien

21 years, 4 months

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Re: [Serusers] FCP support in SER: Modifying SDP

by jaime.gill＠orange.co.uk

Thanks Jiri, I'll have a look at the CVS version. I'm sure there will be more hacking to do only to make it compile.... :( However, got the replacement of the "c=.." part working. I'll test this as soon as I can. Jaime Jiri Kuthan <jiri(a)iptel.org> on 24/02/2003 13:20:00 To: Jaime GILL/EN/HTLUK@HTLUK Nils Ohlmeier <nils(a)ohlmeier.de> cc: Jan Janak <J.Janak(a)sh.cvut.cz> serusers(a)lists.iptel.org Subject: Re: [Serusers] FCP support in SER: Modifying SDP At 12:50 PM 2/24/2003, jaime.gill(a)orange.co.uk wrote: [...] >It think the problem is in the replacement of the SDP information. The first >occurrence of the IP address in "v= " and the port in "m= " in the SDP get >replaced, but the second IP in "c=" is not. nit: it's not the "v=" line, but "o=" ('owner') line which you are replacing. However, you are not probably worried so much about this one -- it maintains primarily a (not widely utilizied) identification purpose. All "c=" occurences do matter. (In addition to port numbers in "m=" lines.) [...] >I have been trying to understand how the proxy builds the forwarded message from >the old one, and realised that for the Via replacement (or adding of more >params), I need to be using a string called add_to_branch_s and >add_to_branch_len (so ignore the replace_via implementation in the current >tar.gz). I suggest you used the mhomed option (available only on CVS). The issue is you need to print the correct IP address in Via on multihomed host. With mhomed enabled, IP routing is utilized to determine the right IP address. Let me know if you need something more for getting Via right. >But for the SDP, whenever I work with get_body, it does not modify it >appropriately. So currently, I'm using msg->orig to get to the initial message, >search for certain IP4 and audio strings and replace them with the information >provided by the fcp server. That means, in the case of the SDP, 2 IP address >replacements (in v=.. and c=..) and 1 port replacement (in m=..). As I mentioned >before, I only manged to change the v=.. and m=... Whenever I try to replace >more than one appearance, strange things happen, like strings in non expected >places, like Via, and cannot work out why. So my question is an open one:): what >is the best way to change the SDP part? I suggest here too -- use the CVS version. It has departed from the use of the buffers (orig and buf) -- we have now just one buffer (buf) without any zero termination. Previously, the two buffers and 0-termination caused lot of issues, some of them possibly annoying you right now. Look at the textops/replace_all action (only on CVS too) to see how to replace multiple occurences of a string in SIP messages. (Caution: you will eventually need to calculate new SDP body size and change content-length too.) >The other of my questions is whether all this mess with NAT's will get solved >when the proxy supports TCP, The major problem is media, which will keep using UDP. >and whether this is the best approach to solve the >SIP through NAT/FW problem. As all NAT traversal methods -- none of them is perfect, each has cons and pros. The benefit of FCP is that once fcpd works, maintenance of the SIP code is easier in user space. Also, you can better couple your pinhole policy with SER's SIP-layer policy. >For example, how about a nathelper module for >netfilter/iptables that gets this working, in the same manner as IRC or ftp >currently? Does anybody know about any work progressing this for linux/FreeBSD? I'm not aware of such. There is Billy Biggs masquerading module, but it is pretty old and no longer maintained. -Jiri _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

21 years, 4 months

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Re: [Serusers] FCP support in SER: Modifying SDP

by jaime.gill＠orange.co.uk

Hi Nils, Once again I'm writing to ask about more things and to give you an update on what is happening with the FCP tests... We have set up a simple scenario with 2 proxies separated by a firewall/NAT, and 2 UA, one within (UA1) and another outside the firewall (UA2). One UA registers with the natted proxy (UA1), the other with the "public" proxy (UA2). At the moment, SIP messages go forwards and backwards without problems, but media is not flowing across the firewall. It think the problem is in the replacement of the SDP information. The first occurrence of the IP address in "v= " and the port in "m= " in the SDP get replaced, but the second IP in "c=" is not. I have been trying all sorts of things, but no joy :( . Here are the INVITE messages in more detail. I also include the latest fcp module for you to play with it. ---------- UA1 to proxy message ------------------ U 172.21.68.78:1129 -> 192.168.6.153:5060 INVITE sip:jaime@asereje.orange.co.uk SIP/2.0 Call-ID: 5812832001907970791(a)172.21.68.78 Content-Length: 121 Content-Type: application/sdp To: sip:jaime@asereje.orange.co.uk From: sip:pepe@asereje.orange.co.uk;tag=-779729009 Contact: sip:pepe@172.21.68.78:5061 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.21.68.78:5061;branch=AC15444E13C5000000F38F819DE5-2*0 v=0..o=- 1046084768435 1046084768465 IN IP4 172.21.68.78 s=- c=IN IP4 172.21.68.78 t=0 0 m=audio 5006 RTP/AVP 8 3 0 ---------- End of UA1 to proxy message ----------------- -------- Proxy to UA2 message --------------------- U 192.168.6.153:5060 -> 172.21.68.78:15592 INVITE sip:172.21.68.78:15592 SIP/2.0 Call-ID: 5812832001907970791(a)172.21.68.78 Content-Length: 121 Content-Type: application/sdp To: sip:jaime@asereje.orange.co.uk From: sip:pepe@asereje.orange.co.uk;tag=-779729009 Contact:<sip:192.168.0.1:33240> CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.6.153;branch=z9hG4bK1019.21c52996.0 Via: SIP/2.0/UDP 172.21.68.78:5061;branch=AC15444E13C5000000F38F819DE5-2*0 v=0..o=- 1046084768435 1046084768465 IN IP4 192.168.0.1 s=- c=IN IP4 172.21.68.78 <--- Need to change this as well!!! t=0 0 m=audio 33240 RTP/AVP 8 3 0 ------------ End of Proxy to UA2 message ------------------- (See attached file: fcp-module210203.tar.gz) I have been trying to understand how the proxy builds the forwarded message from the old one, and realised that for the Via replacement (or adding of more params), I need to be using a string called add_to_branch_s and add_to_branch_len (so ignore the replace_via implementation in the current tar.gz). But for the SDP, whenever I work with get_body, it does not modify it appropriately. So currently, I'm using msg->orig to get to the initial message, search for certain IP4 and audio strings and replace them with the information provided by the fcp server. That means, in the case of the SDP, 2 IP address replacements (in v=.. and c=..) and 1 port replacement (in m=..). As I mentioned before, I only manged to change the v=.. and m=... Whenever I try to replace more than one appearance, strange things happen, like strings in non expected places, like Via, and cannot work out why. So my question is an open one:): what is the best way to change the SDP part? The other of my questions is whether all this mess with NAT's will get solved when the proxy supports TCP, and whether this is the best approach to solve the SIP through NAT/FW problem. For example, how about a nathelper module for netfilter/iptables that gets this working, in the same manner as IRC or ftp currently? Does anybody know about any work progressing this for linux/FreeBSD? Greetings, Jaime Nils Ohlmeier <nils(a)ohlmeier.de> on 18/02/2003 02:58:47 To: Jaime GILL/EN/HTLUK@HTLUK cc: Jan Janak <J.Janak(a)sh.cvut.cz> Jiri Kuthan <jiri(a)iptel.org> Subject: Re: [Serusers] FCP support in SER: Modifying SDP Hi Jaime, debugging without the code is really hard :-) But maybe your problme with SDP is correlated to a bug in the Via header which i marked below. Greetings Nils On Monday 17 February 2003 11:58, jaime.gill(a)orange.co.uk wrote: > U 192.168.6.153:5060 -> 172.21.68.78:5061 > INVITE sip:pepe@172.21.68.78:5061 SIP/2.0..Via: SIP/2.0/UDP > 192.168.6.153;b ranch=z9hG4bKb848.8a014f84.0..Via: SIP/2.0/UDP > 192.168.0.1192.168.0.1:9439. .From: "jaime" ^^^^^^^^^^^^^^^^^^^^^^ Here you inserted the external IP twice. Maybe your SDP replacer did this? > <sip:jaime@asereje.orange.co.uk>;tag=8c20540f-4259-11d7-9cc5 > -00065b4c11cb..To: <sip:pepe@asereje.orange.co.uk>..Call-ID: 8c205410-4259- > 11d7-9cc5-00065b4c11cb@172.21.68.78..CSeq: 1 INVITE..Contact:<sip:192.168.0 > .1:33186>.User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Conte > nt-Length: 211....v=0..o=gill_j 0 0 IN IP4 172.21.68.78..s=session..c=IN IP > 4 172.21.68.78..b=CT:1000..t=0 0..m=audio 33186 RTP/AVP 97 0 8 4..a=rtpmap: > 97 red/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/80 > 00.. -- gpg-key: http://www.ohlmeier.org/public_key.asc ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

21 years, 4 months

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[Serusers] Cannot add "alias" mysql DB (using default passwords)

by Lenny Tropiano

I have turned on mysql authentication, and was able to add users, but aliases complain that "table" doesn't exist: $ serctl add lenny abcdef lenny(a)voiping.com MySql Password: new user added $ serctl alias add 1234 sip:lenny@voiping.com sip:lenny@voiping.com 400 Table 'aliases' Not Found $ mysql -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 243 to server version: 3.23.53 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql> connect ser; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Connection id: 244 Current database: ser mysql> show tables; +-----------------+ | Tables_in_ser | +-----------------+ | acc | | active_sessions | | aliases | | config | | event | | grp | | location | | missed_calls | | pending | | phonebook | | reserved | | silo | | subscriber | | version | +-----------------+ 14 rows in set (0.00 sec) mysql> --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

21 years, 4 months

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[Serusers] On CVS, Radius and Generally SER Status (2003/II)

by Jiri Kuthan

I'm observing increasing number of downloads from CVS and would like to issue few disclaimers to prevent any disappointments. The CVS code is work in progress -- it has not been integrated, tested, it is not complete and we cannot provide help on incomplete software. Also, folks should know that compilation of one of the most frequently asked features, Radius support, is disabled on purpose. It is an external contribution which deserves a clean-up, we are working on it but we are not there yet. My previous forecast for the upcoming release is turning out to be too optimistic. We decided to carry out some more clean-ups and the work takes time. Anyway, we think it is better to clean up now than later -- interest in ser is growing, and we need to keep the codebase sane despite all the new features. -Jiri -- Jiri Kuthan http://iptel.org/~jiri/

21 years, 4 months

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[Serusers] problem in using acc.so and radius_acc.so and auth.so

by Benny Ho

Dear all, after days of trial, I finally load radius_acc.so in my ser. But when I testing the modules with the new routing config. Nothing happen. I follow the sample config in the CVS as follow. I check the radius server with radius -x, no packet seem to send to radius server. I also check the Mysql database, no entry is wrote to acc table. Why? Can anyone tell me what's wrong? ================== [benny@md benny]$ more /etc/ser/ser.cfg # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" [benny@md benny]$ cat /etc/ser/ser.cfg # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" # Uncomment this to use radius for accounting loadmodule "/usr/lib/ser/modules/radius_acc.so" loadmodule "//usr/lib/ser/modules/acc.so" #--- radius_ acc.so params --- modparam("radius_acc","log_level",1) modparam("radius_acc","acc_flag",1) modparam("radius_acc","report_ack",1) modparam("radius_acc","early_media",0) modparam("radius_acc","failed_transactions",0) #--- acc.so params --- modparam("acc","log_level",1) modparam("acc","acc_flag",1) modparam("acc","report_ack",1) modparam("acc","early_media",1) modparam("acc","failed_transactions",1) loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") #modparam("auth", "calculate_ha1", yes) #modparam("auth", "calculate_ha1", no) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present # rewriteFromRoute(); #if (method=="INVITE") { addRecordRoute(); #} else { rewriteFromRoute(); #}; # 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") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #labeled all transaction for accouting setflag(1); #record-route INVITES to make sure BYEs will visit our server too #if (method=="INVITE") addRecordRoute(); # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } [benny@md benny]$ ==================================

21 years, 4 months

2
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[Serusers] NAT Traversal Technique Survey

by Jiri Kuthan

Folks, please help me -- share with me techniques for NAT traversal you use and have hands-on experience with. People repeatedly ask about it, and I'd like to create an FAQ that reflects deployment experience and as wide user feed-back as possible. Just tell me the technique you use, its requirements, limitations, the devices it is known (not) to work with, why you prefer one method over the other, etc. I'll then try to compile it in an FAQ. So please send me an e-mail, an example is attached. I will appreciate any practical details. Thank you, -Jiri ---------------------------------------------------------------- technique: using symmetric communication requirements: phone devices that support symmetric communication; existing species: Cisco's ATA configuration practice: ATAs need to be configured to advertise public address in signaling, or learn it from REGISTER replies; alternatively, one can rewrite signaling using ser's nethelper module; one needs to rewrite SIP anyway because ATAs don't advertise their symmetricity; see www.foo.bar for info on configuring ATA... limitations: non-symmetric devices, like Messenger don't work; misc: ATA has no display, that's why I am anxiously waiting for more vendors to support symmetric signaling ---------------------------------------------------------------- technique: UPnP requirements: NATs and phones with UPnP support; Messenger and snom are known to support UPnP; there is linux support for it configuration practice: of course, upnp requires by definition no configuration ;-) (I'm not serious -- anyone actually tried it?) ---------------------------------------------------------------- technique: geek tweaks: set-up port forwading manually configuration practice: you need to configure NATs to split its public-side port numbers accross your private-side phones, and configure the phones (if they allows so) to use these port numbers; also, phones need to be configured to use publicly reachable address in their payloads requriements: configurable NATs (many residental NATs are configurable) and configurable phones (ATAs do that, I heard pingtel did it too) ---------------------------------------------------------------- technique: ALG requirements: SIP-capable NAT (like Intertex or Cisco/PIX) issues: intertex freezes my ssh connections after some time on-line and elderly models don't like all Ethernet devices; when things don't work, the red-button off-on helps sometimes ---------------------------------------------------------------- technique: STUN requirements: STUN-enabled phone (like k-phone, snom) limitations: doesn't work over symmetric NATs (words-of-mouth propaganda has been telling me that many residential NATs are fortunately not symmetric, but I don't know how objective this information really is) ---------------------------------------------------------------- -- Jiri Kuthan http://iptel.org/~jiri/

21 years, 4 months

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[Serusers] Windows- Sms gateway

by ant buckland

Hi- I am a very Junior designer, and as part of my efforts to impress my boss, I am trying to create a simple email to sms application. Sadly, here in South Africa, the cellular service providers do not provide an sms gateway to the public, and so I am researching the iptel.org SIP Express Router. I need to know the following: after reading the iptel.org website, I would like to know whether it is possible to install the application on a Windows 2000 server, and run the gateway. From what I read, apparently only the core application can be installed... and this does not appear to include the gateway? Obviously, I will pass the project on to a programmer who knows more, so you will not be plagued by too-simple questions from an inexperienced designer... but I would appreciate any assistance/info that will allow me to convince my boss. Thank you! ph (021) 685 6499 anthony_buckland(a)yahoo.co.uk 70 Strubens Rd, Observatory, Cape Town --------------------------------- With Yahoo! Mail you can get a bigger mailbox -- choose a size that fits your needs

21 years, 4 months

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[Serusers] Absorbing provisional responses

by Sebastien Pierre

Hi all, This message is somewhat related to a preceding one (Simple SER Use Case), because I realised I did not ask the right question. Here it is: I want to make a proxy that will try to locate the callee by sequentially trying n different UA, by sending an INVITE and waiting for either OK or timeout to occur. On timeout, the proxy will try the next UA. The detail here is that the proxy sends a "100 Trying" to the caller *before* contacting the various UA, and that *no more* "100 Trying" are forwarded by the proxy, which absorbs those responses from contacted UA. Jiri indicated that the configuration presented in <http://www.iptel.org/ser/doc/seruser-html/x510.html#AEN616> realised the scenario I was mentioning... but it seems to me that the proxy does not send and then absorb "100" responses, but rather relays this responses to the caller. This would lead to the caller receiving as many "100" responses as contacted UAs. Is it possible to do this with SER ? TIA, -- Sebastien

21 years, 4 months

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[Serusers] Crash in SER

by Maxim Sobolev

Hi, We've observed the following crash in ser. Do you have any ideas what the cause could be? I'm keeping corefile and binary compiled with debug symbols, so that don't hesitate to contact me if any additional information is necessary. Thank you in advance! -Maxim Script started on Fri Feb 21 05:43:09 2003 bash-2.05a$ gdb ~/sip/ser/work/ser-0.8.10/ser ser.core GNU gdb 4.18 (FreeBSD) Copyright 1998 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-unknown-freebsd"...Deprecated bfd_read called at /usr/src/gnu/usr.bin/binutils/gdb/../../../../contrib/gdb/gdb/dbxread.c line 2627 in elfstab_build_psymtabs Deprecated bfd_read called at /usr/src/gnu/usr.bin/binutils/gdb/../../../../contrib/gdb/gdb/dbxread.c line 933 in fill_symbuf Core was generated by `ser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /usr/lib/libc.so.4...done. Reading symbols from /usr/local/lib/ser/modules/sl.so...done. Reading symbols from /usr/local/lib/ser/modules/tm.so...done. Reading symbols from /usr/local/lib/ser/modules/rr.so...done. Reading symbols from /usr/local/lib/ser/modules/maxfwd.so...done. Reading symbols from /usr/local/lib/ser/modules/usrloc.so...done. Reading symbols from /usr/local/lib/ser/modules/registrar.so...done. Reading symbols from /usr/local/lib/ser/modules/nathelper.so...done. Reading symbols from /usr/local/lib/ser/modules/textops.so...done. Reading symbols from /usr/local/lib/ser/modules/radius_auth.so...done. Reading symbols from /usr/local/lib/libradiusclient.so.0...done. Reading symbols from /usr/lib/libcrypt.so.2...done. Reading symbols from /usr/libexec/ld-elf.so.1...done. #0 0x807bdd3 in qm_status (qm=0x2816d000) at mem/q_malloc.c:447 447 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++); (gdb) bt #0 0x807bdd3 in qm_status (qm=0x2816d000) at mem/q_malloc.c:447 #1 0x8079a70 in qm_debug_frag (qm=0x2816d000, f=0x282fd0a4) at mem/q_malloc.c:109 #2 0x807ab37 in qm_free (qm=0x2816d000, p=0x282fd0bc, file=0x2a197a2d "t_reply.c", func=0x2a197e28 "t_on_reply", line=826) at mem/q_malloc.c:359 #3 0x2a188000 in t_on_reply (p_msg=0x80dfe30) at t_reply.c:826 #4 0x805933a in forward_reply (msg=0x80dfe30) at forward.c:346 #5 0x806af1c in receive_msg (buf=0x80c3340 "SIP/2.0", len=360, src_su=0x81dafa0) at receive.c:161 #6 0x8077ff0 in udp_rcv_loop () at udp_server.c:336 #7 0x805eaa5 in main_loop () at main.c:618 #8 0x806197d in main (argc=1, argv=0xbfbffc04) at main.c:1383 #9 0x804c5bd in _start () (gdb) up #1 0x8079a70 in qm_debug_frag (qm=0x2816d000, f=0x282fd0a4) at mem/q_malloc.c:109 109 qm_status(qm); (gdb) l 104 }; 105 if ((FRAG_END(f)->check1!=END_CHECK_PATTERN1)|| 106 (FRAG_END(f)->check2!=END_CHECK_PATTERN2)){ 107 LOG(L_CRIT, "BUG: qm_*: fragm. %p end overwritten(%x, %x)!\n", 108 f, FRAG_END(f)->check1, FRAG_END(f)->check2); 109 qm_status(qm); 110 abort(); 111 } 112 if ((f>qm->first_frag)&& 113 ((PREV_FRAG_END(f)->check1!=END_CHECK_PATTERN1) || (gdb) up #2 0x807ab37 in qm_free (qm=0x2816d000, p=0x282fd0bc, file=0x2a197a2d "t_reply.c", func=0x2a197e28 "t_on_reply", line=826) at mem/q_malloc.c:359 359 qm_debug_frag(qm, f); (gdb) l 354 return; 355 } 356 prev=next=0; 357 f=(struct qm_frag*) ((char*)p-sizeof(struct qm_frag)); 358 #ifdef DBG_QM_MALLOC 359 qm_debug_frag(qm, f); 360 if (f->u.is_free){ 361 LOG(L_CRIT, "BUG: qm_free: freeing already freed pointer," 362 " first free: %s: %s(%d) - aborting\n", 363 f->file, f->func, f->line); (gdb) up #3 0x2a188000 in t_on_reply (p_msg=0x80dfe30) at t_reply.c:826 826 shm_free(ack); (gdb) l 821 /* acknowledge negative INVITE replies */ 822 if (t->is_invite && (msg_status>=300 || (t->local && msg_status>=200))) { 823 ack = build_ack( p_msg, t, branch , &ack_len); 824 if (ack) { 825 SEND_PR_BUFFER( &uac->request, ack, ack_len ); 826 shm_free(ack); 827 } 828 } /* ack-ing negative INVITE replies */ 829 830 /* clean-up the transaction when transaction completed */ (gdb) up #4 0x805933a in forward_reply (msg=0x80dfe30) at forward.c:346 346 if (mod->exports->response_f(msg)==0) goto skip; (gdb) l 341 /* quick hack, slower for mutliple modules*/ 342 for (mod=modules;mod;mod=mod->next){ 343 if ((mod->exports) && (mod->exports->response_f)){ 344 DBG("forward_reply: found module %s, passing reply to it\n", 345 mod->exports->name); 346 if (mod->exports->response_f(msg)==0) goto skip; 347 } 348 } 349 350 /* we have to forward the reply stateless, so we need second via -bogdan*/ (gdb) print msg $1 = (struct sip_msg *) 0x80dfe30 (gdb) print msg [1@* $2 = {id = 1104, first_line = {type = 2, u = {request = {method = {s = 0x80c3340 "SIP/2.0", len = 7}, uri = {s = 0x80c3348 "488", len = 3}, version = { s = 0x80c334c "Not Acceptable Here", len = 19}, method_value = 488}, reply = {version = { s = 0x80c3340 "SIP/2.0", len = 7}, status = {s = 0x80c3348 "488", len = 3}, reason = { s = 0x80c334c "Not Acceptable Here", len = 19}, statuscode = 488}}}, via1 = 0x80e27fc, via2 = 0x80e2d1c, headers = 0x80e620c, last_header = 0x80e0b08, parsed_flag = 63, h_via1 = 0x80e620c, h_via2 = 0x80e0b98, callid = 0x80e0008, to = 0x80e10ac, cseq = 0x80e0b08, from = 0x80e0f2c, contact = 0x0, maxforwards = 0x0, route = 0x0, record_route = 0x0, content_type = 0x0, content_length = 0x0, authorization = 0x0, expires = 0x0, proxy_auth = 0x0, www_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, eoh = 0x0, unparsed = 0x80c3493 "Content-Length: 0\r\n\r\n", src_ip = { af = 2, len = 4, u = {addr32 = {1214690368, 0, 0, 0}, addr16 = {46144, 18534, 0, 0, 0, 0, 0, 0}, addr = "@╢fH", '\000' <repeats 11 times>}}, src_port_no = 5061, dst_ip = {af = 2, len = 4, u = {addr32 = {1214690368, 0, 0, 0}, addr16 = {46144, 18534, 0, 0, 0, 0, 0, 0}, addr = "@╢fH", '\000' <repeats 11 times>}}, orig = 0x80e2a48 "SIP/2.0 488 Not Acceptable Here\r\nVia: SIP/2.0/UDP 64.180.102.72:5060;branch=z9hG4bKfa9f.17ad3774445fef6b70f18b74de26e0b2.0\r\nVia: SIP/2.0/UDP 172.17.1.127:5060;rport=5060\r\nTo: <sip:151@64.180.102.72;us"..., buf = 0x80c3340 "SIP/2.0", len = 360, new_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0}, add_rm = 0x0, repl_add_rm = 0x80e0514, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 54 times>, add_to_branch_len = 0, hash_index = 0, flags = 0} (gdb) up #5 0x806af1c in receive_msg (buf=0x80c3340 "SIP/2.0", len=360, src_su=0x81dafa0) at receive.c:161 161 forward_reply(msg); (gdb) l 156 gettimeofday( & tvb, &tz ); 157 STATS_RX_RESPONSE ( msg->first_line.u.reply.statuscode / 100 ); 158 #endif 159 160 /* send the msg */ 161 forward_reply(msg); 162 163 #ifdef STATS 164 gettimeofday( & tve, &tz ); 165 diff = (tve.tv_sec-tvb.tv_sec)*1000000+(tve.tv_usec-tvb.tv_usec); (gdb) q bash-2.05a$ exit Script done on Fri Feb 21 05:44:36 2003

21 years, 4 months

2
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[Serusers] problem with loading compiled radius modules

by Benny Ho

I had the same error msgs when I load the radius_acc modules the following is the error msgs [root@md benny]# /usr/sbin/ser restart 0(3699) ERROR: load_module: could not open module </usr/lib/ser/modules/radius_ acc.so>: libradiusclient.so.0: cannot open shared object file: No such file or d irectory 0(3699) parse error (25,13-48): failed to load module ERROR: bad config file (1 errors) [root@md benny]# and I check that the radius_acc.so is in /usr/lib/ser/modules and the libradiusclient.so.0 is also in /usr/local/lib as it is stated in the Makefile. Where is the error came from? Have any one successfully implement it?

21 years, 4 months

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[Serusers] RE: RE: problem with loading compiled radius modules

by Yang Xiang

Hi Benny, I had the same problem as you. After I statically linked the radiusclient library in the both radius modules this problem seems to be resolved. http://lists.iptel.org/pipermail/serusers/2003-February/000419.html However radius authentication still doesn't work and the developer said: "the path is not staticaly linked, it should be resolved by you dynamic linker, I don't know where is the problem, it seems to me that there is some problem with radiusclient libraries on your system, " I use Solaris 8. Now I am trying to recompile and reinstall radiusclient library. If I find out something, I'll tell you. regards, yang

21 years, 4 months

1
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[Serusers] CVS version current, gives this error

by Lenny Tropiano

Default config (ser.cfg installed from gmake install), gives this error (I bumped up the debug level and had it log to stderr): p2:[root@enterprise.icus.com] 31 $ ser 0(62901) loading module /usr/local/lib/ser/modules/sl.so 0(62901) loading module /usr/local/lib/ser/modules/tm.so 0(62901) loading module /usr/local/lib/ser/modules/rr.so 0(62901) loading module /usr/local/lib/ser/modules/maxfwd.so 0(62901) loading module /usr/local/lib/ser/modules/usrloc.so 0(62901) loading module /usr/local/lib/ser/modules/registrar.so 0(62901) find_param_export: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(62901) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: <rewriteFromRoute> not found 0(62901) parse error (80,19-20): unknown command, missing loadmodule? 0(62901) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(62901) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(62901) INFO: signal 15 received 0(62901) Memory status (pkg): 0(62901) qm_status (0x80a1780): 0(62901) heap size= 1046128 0(62901) used= 0, used+overhead=2448, free=1043680 0(62901) max used (+overhead)= 2448 0(62901) dumping all allocked. fragments: 0(62901) dumping free list stats : 0(62901) hash= 137. fragments no.: 1 0(62901) ----------------------------- --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

21 years, 4 months

2
2
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[Serusers] Some compile time errors for FreeBSD of CVS Tree source (sip_router)

by Lenny Tropiano

pass_fd.c: added #include <sys/uio.h> tcp_main.c: moved #include <sys/select.h> underneath #include <sys/types.h.> Still getting this: $ gmake Makefile.defs:331: Old gcc detected (2.9x), use gcc >= 3.1 for better results gcc -g -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.11pre6-tcp5-tm"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 2.95"' -D__CPU_i386 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DVOICE_MAIL -D_TOTAG -DUSE_TCP -DFAST_LOCK -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -c tcp_main.c -o tcp_main.o tcp_main.c: In function `tcp_send': tcp_main.c:351: `MSG_NOSIGNAL' undeclared (first use in this function) tcp_main.c:351: (Each undeclared identifier is reported only once tcp_main.c:351: for each function it appears in.) gmake: *** [tcp_main.o] Error 1 Cannot find proper #define for MSG_NOSIGNAL The flags, according to "send()" man page is: The flags parameter may include one or more of the following: #define MSG_OOB 0x1 /* process out-of-band data */ #define MSG_PEEK 0x2 /* peek at incoming message */ #define MSG_DONTROUTE 0x4 /* bypass routing, use direct interface */ #define MSG_EOR 0x8 /* data completes record */ #define MSG_EOF 0x100 /* data completes transaction */ The flag MSG_OOB is used to send ``out-of-band'' data on sockets that support this notion (e.g. SOCK_STREAM); the underlying protocol must also support ``out-of-band'' data. MSG_EOR is used to indicate a record mark for protocols which support the concept. MSG_EOF requests that the sender side of a socket be shut down, and that an appropriate indication be sent at the end of the specified data; this flag is only implemented for SOCK_STREAM sockets in the PF_INET protocol family, and is used to implement Transaction TCP (see ttcp(4)). MSG_DONTROUTE is usually used only by diagnostic or routing programs. --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

21 years, 4 months

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[Serusers] Via tags question...

by lgfausak＠august.net

This is a protocol question, I don't know if it is directly related to ser. Concerning the Via tags, should any proxy/gateway modify tags that don't belong to it, or should they simply be copied verbatum? For example, I have a situation where I send an INVITE message, and the TRYING response I receive modifies one of the Via: tags. I don't think this is right. Here are the two packets in question: # U 2003/02/19 18:36:20.276844 164.9.42.25:5060 -> 184.294.324.2:5060 INVITE sip:12143357976@184.294.324.2:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:12143357976@164.9.42.25;branch=0>. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. CSeq: 2 INVITE. Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Contact: <sip:8178863582@216.87.145.22:4060>. User-Agent: FXS_GW (sip1028a.bin). Content-Type: application/SDP. Content-Length:291. P-hint: LD GATEWAY. . v=0. o=FXS_GW 12367 0 IN IP4 216.87.145.22. s=Audio Session. i=Audio Session. c=IN IP4 216.87.145.22. t=0 0. m=audio 16384 RTP/AVP 0 18 4 8 96. a=fmtp:96 0-11. a=rtpmap:0 PCMU/8000/1. a=rtpmap:18 G729/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:96 telephone-event/8000. And your response is: # U 2003/02/19 18:36:20.902858 184.294.324.2:5060 -> 164.9.42.25:5060 SIP/2.0 100 Trying . Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone;tag=27DC1DCC-F61. Date: Thu, 20 Feb 2003 00:36:20 GMT. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Content-Length: 0. . See how the : Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Has been changed to: Via: SIP/2.0/UDP 216.87.145.22:4060. Does anyone have any perl script to take a ngrep protocol dump and print it out in a pretty way? ---greg Greg Fausak August.Net Services, LLC

21 years, 4 months

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[Serusers] Add a user to subscriber table.

by Ivan Vignola

Hi, I would add a user to subsriber table by sql command in my application (not using serctl script). What text strings I must use to calculate HA1 (=phplib_id) and ha1b values if USER_ID=ivan,PASSWORD=ivanpswd and REALM=192.168.11.237 ? Thanks in advance. Regards, Ivan Vignola. _________________________________________________________________ Vinci la nuova Nissan Micra con MSN Messenger! http://www.msn.it/messenger/

21 years, 4 months

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[Serusers] serctl - tail problems on solaris8?

by Kamil Ince

bet i missed reading something, again :-/ but wasting too much time on this.. i get on solaris 8 the following: # echo $SIP_DOMAIN myser.my.net # echo $PW heslo # serctl add user1 hispw user1(a)my.net usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] new user added # serctl passwd user1 newpw usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] non-existent user of course the user has been added to the database... anyone came accross that before? tia -k

21 years, 4 months

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[Serusers] [bug] TM module config

by Sebastien Pierre

Hi again, Here is how to get a segfault: <snip> #Load modules required by a stateful User Agent Service loadmodule "modules/tm.so" #SER timer parameters modparam("tm", "fr_inv_timer", "10"); modparam("tm", "fr_timer", "10"); </snip> This happened after quoting the numbers in the modparam. Tested with ser-0.8.10 on Linux RH 8.0 -- Sebastien

21 years, 4 months

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[Serusers] Logging with SER

by Sebastien Pierre

Hi all, I finally found solutions to my problem: 1) It is easier when setting "fork=no" because then ser does not daemonise and stderr is then attached to the terminal in which I started ser 2) The configuration file should set a debugging level ("set debug=1"), and use a number equal or lower to the debug level in the log, like 'log(0, "Hello World")' So my HelloWorld.cfg is as follows: <snip> debug=0 log_stderror=yes fork=no route[0] { log(0, "Hello World!"); } </snip> And it works with "ser -f HelloWorld.cfg" Cheers, -- Sebastien

21 years, 4 months

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Re: [Serusers] SER logging (and the mighty HelloWorld.cfg)

by Sebastien Pierre

Hi Nils! Le mar 18/02/2003 à 16:56, Nils Ohlmeier a écrit : > Hard to say without seeing the config file. But '-E' as command line > parameter should work and will overwrite any value in your config > file. Here it is (the HelloWorld.cfg): <snip> log_stderror=yes route[0] { log("Hello World!"); } </snip> Here is the SIP message I send to SER: <snip> INVITE sip:alice@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5070 To: Alice <sip:alice@localhost> From: Me <sip:me@localhost> Call-ID:123456789@127.0.0.1 CSeq: 1 INVITE Content-Length: 0 </snip> Here is the answer I get from SER: <snip> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.0.1:5070 To: Alice <sip:alice@localhost>;tag=d907c037823644515dfe0ede38ca9976.de25 From: Me <sip:me@localhost> Call-ID:123456789@127.0.0.1 CSeq: 1 INVITE Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=9995 req_src_ip=127.0.0.1 in_uri=sip:alice@127.0.0.1 out_uri=sip:alice@127.0.0.1 via_cnt==1" </snip> And here is what I get from SER in stdout/stderr: <snip> $ ser -E HelloWorld.cfg Listening on 127.0.0.1 [127.0.0.1]::5060 172.25.49.134 [172.25.49.134]::5060 Aliases: XXX.alcatel.fr:5060 localhost:5060 localhost.localdomain:5060 XXX:5060 </snip> And from /var/log/messages <snip> Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is initially 65535 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is finally 131070 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is initially 65535 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is finally 131070 Feb 18 17:33:21 frilld0030711 ser[10003]: INFO: fifo process starting: 10003 Feb 18 17:33:21 frilld0030711 ser[10003]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... </snip> So there is sadly no "HelloWorld!" somewhere... would it be because of the "404 not found" error I get ? TIA, -- Sebastien

21 years, 4 months

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[Serusers] Simple SER Use Case

by Sebastien Pierre

Hi all, I am trying to implement a very simple call forwarding use case with SER. This use case is directly taken from a book entitled "Internet Applications with SIP". Here are the actors: - SIP caller, wants to call <alice@domain> - Proxy server of <domain> - SIP phone which is default terminal for <alice(a)domain.com> - Voicemail server Here is the scenario: 0) SIP caller INVITEs <alice@domain> 1) Proxy receives INVITE 2) Proxy forwards INVITE to alice default SIP phone 3) Proxy responses 100 TRYING to SIP caller 4) SIP phone responses 100 TRYING to proxy 5) SIP phone responses 180 RINGING to proxy 6) Proxy reponses 180 RINGING to caller 7) After timeout, proxy send CANCEL to default SIP phone 8) SIP phone responses OK to proxy... ... X) Proxy forward original INVITE to voice mail server... The fact is that the proxy first notifies the caller that it is trying to locate alice, listens for "180 RINGING" and forward it to the caller, then cancels on the SIP phone side because of a timeout -- then the proxy forwards the INVITE to the voice mail. I have tried some configurations, but I did not see how to listen for the 180 RINGING and forward them to the caller... is there a way to bind a "route" to a provisional response like: route[0] { ... #In case a 180 occur, we execute route[1] t_on_provisional("180", "1"); #But we're also waiting for an OK or a timer #event to arrive t_on_negative("2"); t_relay(); ... } route[1] { #Forwards the 180 Ringing from the SIP phone to #the caller t_relay(); } BTW, I was wondering if t_on_negative is a *blocking* action ? TIA, -- Sebastien

21 years, 4 months

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[Serusers] ser-mysql for solaris

by Kamil Ince

hello, can someone help me with the database details for ser? I got only solaris available, and couldnt find ser-mysql as "non-rpm" tia -k

21 years, 4 months

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[Serusers] Re: [Serhelp] SER logging

by Sebastien Pierre

Hi Nils! Le mar 18/02/2003 à 16:03, Nils Ohlmeier a écrit : > On Tuesday 18 February 2003 15:57, Sebastien Pierre wrote: > > I was just wondering (did not find in User or Dev guide) where to get > > the result of SER "log" action... I have written a fake SIP UA that > syslog (probably /var/log/messages) if log_stderror=no, otherwise stderr (your > console). I thought I did not install syslog with root privileges, so I did not look in the *obvious* file! Anyway, I put a "log_sterror=yes" line at the beginning of my SER configuration script, but SER keeps outputting on /var/log/messages and not on STDERR... what did I do wrong ? > > sends SIP messages written in a file to test custom SER configurations, > > but I don't know if SER properly receives my messages or executes my > > routing program. Any advices ? > Congratulation for reinventing the wheel -> http://sipsak.berlios.de :-) I know about it but... >> sipsak -s sip:alice@localhost error: this FQDN or IP is not valid: frwed00304 I'm still puzzled about this FQDN and my hostname... and I could not resist making a 20 lines Python script to get SER tested ;) -- Seb

21 years, 4 months

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Re: [Serusers] SerWeb

by Karel Kozlik

Hello, check your php.ini file if you have: register_globals = On Karel > > >Hi, > > > >I' ve installed SERWEB.....But when I try to register or login...nothing happens... the page is reload. The Apache server is working. Anyone had this same problem? > > > >Thanks, > > > >Adriana Simizo > >Telefonica Brasil - Telecomunicaçőes de Săo Paulo S. A. > > > > -- > Jiri Kuthan http://iptel.org/~jiri/

21 years, 4 months

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Re: [Serusers] problem with loading compiled radius modules

by Xiang

I have resolved this problem. The radiusclient library was not included in the both radius modules, somehow. After I statically linked the radiusclient library with the option -lradiusclient when compiling, it works. regards yang ------------------- > The modules need radiusclient library but linker is unable to find > symbols from the library. Do you have the radiusclient library installed > correctly ? > > regards, Jan. > > On 14-02 14:35, Yang Xiang wrote: > > Hi, > > > > I get the same error. > > > > I have compiled the whole server and try to load modules on the host where > > the server was compiled. Unfortunatelly it doesn't work. I get the same > > error message as before. > > > > What should I do???? > > > > Regards > > > > Yang > > > > > > ----- Original Message ----- > > From: "Jan Janak" <J.Janak(a)sh.cvut.cz> > > To: "Yang Xiang" <yang.xiang(a)iitb.fraunhofer.de> > > Sent: Friday, February 14, 2003 2:03 PM > > Subject: Re: [Serusers] problem with loading compiled radius modules > > > > > > > Recompile the whole server if you can. As you said you use precomplied > > > binaries and it is possible that it was compiled using a different > > > compiler. > > > > > > regards, Jan. > > > > > > On 14-02 13:56, Yang Xiang wrote: > > > > Hi, > > > > > > > > Now I get more error messages: > > > > > > > > 0(22528) ERROR: load_module: could not open module > > > > </usr/local/lib/ser/modules/radius_acc.so>: ld.so.1: ser: fatal: > > relocation > > > > error: file /usr/local/lib/ser/modules/radius_acc.so: symbol > > rc_avpair_add: > > > > referenced symbol not found > > > > 0(22528) parse error (30,13-54): failed to load module > > > > 0(22528) ERROR: load_module: could not open module > > > > </usr/local/lib/ser/modules/radius_auth.so>: ld.so.1: ser: fatal: > > relocation > > > > error: file /usr/local/lib/ser/modules/radius_auth.so: symbol > > > > rc_read_config: referenced symbol not found > > > > 0(22528) parse error (31,13-55): failed to load module > > > > ERROR: bad config file (2 errors) > > > > > > > > What is wrong? In acc_mod.c and auth_mod.c I changed the definition of > > > > CONFIG_FILE and let it point to an absolute path > > > > "/usr/local/etc/radiusclient/radiusclient.conf". By the way, I compiled > > the > > > > modiles on another mashine and copied them to the mashine where ser > > runs. Is > > > > that right? > > > > > > > > regards, > > > > > > > > yang > > > > > > > > > > > > > >

21 years, 4 months

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[Serusers] Is there a tool for benchmarking SER?

by Maxim Sobolev

Folks, Our company mostly finished development of the SIP VoIP system that consists of the SER for registration, de-NATifying, and routing and B2BUA for debit applications. Now we would like to test performance of the whole system under the load. In the SER's documentation I saw very impressive performance numbers and wonder how they were obtained. Are there any freely-available tools for creating huge number of SIP sessions simulateneously and if yes, how they can be obtained? Any pointers are greatly appreciated. Thanks! -Maxim

21 years, 4 months

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[Serusers] calling a module

by Mario Kolberg

Hi, > simply use fi_check() in the configuration script. That will call your > function. Thats what I thought also, but unfortunately, SER does not run when. If I run it with a call fi_check() in the ser.cfg file it logs a parse error: Feb 14 14:15:00 DISCUS ser: parse error (39,1-9): parse error Feb 14 14:15:00 DISCUS ser: parse error (39,1-9): I have attached the complete cfg file. But if I remove that fi_check() call, ser runs fine. Have I messed something up with parameters???? Thanks for your help! Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it. # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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" alias="cs.stir.ac.uk" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #mario loadmodule "//usr/lib/ser/modules/mysql.so" loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" loadmodule "//usr/lib/ser/modules/fi.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #mario loadmodule "//usr/lib/ser/modules/auth.so" fi_check(); # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #mario modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") #mario modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #mario modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present rewriteFromRoute(); # 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") { # Uncomment this if you want to use digest authentication #mario if (!www_authorize("cs.stir.ac.uk", "subscriber")) { www_challenge("cs.stir.ac.uk", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; }

21 years, 4 months

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[Serusers] writing a module

by Mario Kolberg

Hi, I am trying to write a module for SER. I managed to write a tiny test module ("hello world") and it compiles fine (the source is attached). However, I'm not quite sure I fully understood how to execute the functions defined in the module. I included the module to sr_modules.c and also a line 'loadmodule' in the ser.cfg. But how do I actually execute the function in the module every time a (INVITE) message is received by the server? The docs couldn't really help ... Thanks very much! Regards, Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it. #include <stdio.h> #include <string.h> #include <stdlib.h> #include "../../sr_module.h" #include "../../dprint.h" #include "../../error.h" #include "../../ut.h" static int fi_check(struct sip_msg* msg,char* str,char* str2); static int mod_init(void); #ifdef STATIC_FI struct module_exports fi_exports = { #else struct module_exports exports= { #endif "fi_module", (char*[]){ "fi_check" }, (cmd_function[]){ fi_check }, (int[]){ 0 }, (fixup_function[]){ 0 }, 1, NULL, NULL, NULL, 0, mod_init, (response_function) 0, (destroy_function) 0, 0, 0 }; static int mod_init(void) { printf("FI module initializing\n"); return 0; } static int fi_check(struct sip_msg* msg, char* str1,char* str2) { printf("FI module executing\n"); return 1; }

21 years, 4 months

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Re: [Serusers] Routing with ser / Problems with linphone

by Jiri Kuthan

/* digging through the archives to make sure I haven't missed anything while I was off-line last weeks ... */ I actually think that is a backwards compatibility issue in SIP. Loose-routing works correctly if sender knows whether the next-hop is loose or strict router. That is a correct assumption as long as the routing information collected automatically in record-route tells you that. If you use pre-configured Route along with an outbound proxy, you configure this strict/loose on your own and you need to get it right. Hence, when you configure loose routing for use with a strict outbound router, things will fail. Anyway, with lr support in next ser release, it should not matter any more. We've had the same issue with kphone previouly, it works since an option has been introduce to enforce use of strict routing. see http://www.ietf.org/mail-archive/working-groups/sipping/current/msg03451.ht… -Jiri At 11:18 AM 2/5/2003, Klaus Darilion wrote: >Hello! > >I've installed ser (sip:obelix.ict.tuwien.ac.at) und want'ed to use it >with linphone, so I configured linphone to use ser as outbound proxy and >registrar. The registration works fine but not the INVITEs. ser answers >to an INVITE from linphone with a 404 Not Found and I found out that the >problem is the "Route" header in the INVITE from linphone. > >Route: <sip:obelix.ict.tuwien.ac.at;lr> > >When I remove the header from the invite and send it manually (sipsak) >the ser proxy accepts the invite and forwards it. I think ser reacts >wrong because if an RFC3261 proxy gets an request with a route header >which points to itself it should remove the header and forward the >request. > >Than I tried the same with an different proxy in the route header: > >Route: <sip:iptel.org;lr> > >Now, my ser proxy (sip:obelix.ict.tuwien.ac.at) accepts the request and >forwards it to iptel.org, but it rewrites the invite to: >INVITE sip:iptel.org;lr SIP/2.0 > >Thats what an RFC2543 proxy would do, but not an RFC3261 proxy, which >only is allowed to do that if the route header has no "lr" parameter or >if he is responsible for the domain in the request URI. If there is an >"lr" parameter the proxy must not change the request URI. > >If I'm wrong please let me know. > >I use ser 0.8.10 with the standard config file and linphone 0.9.1 & >0.10.0 > >Regards, >Klaus >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

21 years, 4 months

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[Serusers] Testing SER

by Jaime Hemmett

Hello, I installed SER, I have started the services. I haven't set up the mysql part yet, I wanted to test it working at all. With bone phone I have tried connecting but gotten: "java.net.BindException: Address already in use" The SER documentation makes out that SER can be used without the mysql parts to make calls with microsoft messenger > version 4.6 ut I've been unable to connect with this either. What now? Jaime.

21 years, 4 months

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[Serusers] Small patch for sip_dictionary

by Maxim Sobolev

Folks, sip_dictionary buindled with ser doesn't include Vendor-Specific attribute. Attached patch should fix it, it would be nice to have it included into the next release. Thanks! -Maxim $FreeBSD$ --- etc/sip_dictionary 2003/01/27 14:13:18 1.1 +++ etc/sip_dictionary 2003/01/27 14:13:39 @@ -48,6 +48,7 @@ ATTRIBUTE Framed-Route 22 string ATTRIBUTE Framed-IPX-Network 23 ipaddr ATTRIBUTE State 24 string +ATTRIBUTE Vendor-Specific 26 string ATTRIBUTE Session-Timeout 27 integer ATTRIBUTE Idle-Timeout 28 integer ATTRIBUTE Termination-Action 29 integer

21 years, 4 months

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Re: [Serusers] User permission question.

by Ivan Vignola

Hi, thank you for your replies. I need be sure that user 'ivan', with password 'ivanpswd', registers himself typing 'sip:ivan.vignola@192.168.11.237' and not typing 'silvio.berlusconi(a)192.168.11.237'. Regards. Ivan. >From: Jiri Kuthan <jiri(a)iptel.org> >To: "Ivan Vignola" <ivanvignola(a)hotmail.com>, serusers(a)lists.iptel.org >Subject: Re: [Serusers] User permission question. >Date: Fri, 14 Feb 2003 11:45:37 +0100 > >I'm not sure what it means "authenticate by his uri". There are quite >many URIs in each request, which BTW have very low security value. >Can you give more details? > >Thanks, > >-Jiri > >At 10:29 AM 2/14/2003, Ivan Vignola wrote: > >Hi, > > > >in my tests user can authenticate himself using user_id, password and any >sip uri. I need user authenticating himself only by his associated uri. How >can I do that? > > > >Thanks in advance. > > > >Ivan Vignola. > > > > > > > > > > > >_________________________________________________________________ > >Vinci la nuova Nissan Micra con MSN Messenger! >http://www.msn.it/messenger/ > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > >-- >Jiri Kuthan http://iptel.org/~jiri/ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ MSN Foto: condividi, ritocca e stampa le tue foto online http://photos.msn.it

21 years, 4 months

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[Serusers] User permission question.

by Ivan Vignola

Hi, in my tests user can authenticate himself using user_id, password and any sip uri. I need user authenticating himself only by his associated uri. How can I do that? Thanks in advance. Ivan Vignola. _________________________________________________________________ Vinci la nuova Nissan Micra con MSN Messenger! http://www.msn.it/messenger/

21 years, 4 months

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[Serusers] problem with loading compiled radius modules

by Yang Xiang

Hi Jan, I have successfully compiled the radius modules radius_acc.so and radius_auth.so. But when I try to load the both modules ser tells me: ERROR: bad config file (2 errors) I use the solaris binary package from iptel and have compiled myself only the both radius modules. Is there any incompability? How can I get mor debug information? Regards, Yang ----------------------------------- My config file: debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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/radius_acc.so" loadmodule "/usr/local/lib/ser/modules/radius_auth.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") #modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic alias="pix-demo.fhg.de" route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present rewriteFromRoute(); # 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") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; }

21 years, 4 months

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[Serusers] SIP2H323 and H3232SIP

by Benny Ho

Hi, How could I integrate SIP and H323 together? I would like to h323 endpoint call sip endpoint. Is it possible using SER? My config is Cisco AS5300, gnu gatekeeper and SER.

21 years, 4 months

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[Serusers] SerWeb

by Adriana Simizo

Hi, I' ve installed SERWEB.....But when I try to register or login...nothing happens... the page is reload. The Apache server is working. Anyone had this same problem? Thanks, Adriana Simizo Telefonica Brasil - Telecomunicações de São Paulo S. A.

21 years, 4 months

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