- sr-users - kamailio.org

[Serusers] eybeam and ser PA module
by Klaus Darilion 19 Jan '05

19 Jan '05

Hi all! On the sipthat blog (http://sipthat.com/) there is an article about the eyebeam SDK which comes with an PA module for ser. ... > The eyeBeam SDK also comes with a basic Presence Agent module for > SER [SIP express Router by iptel.org]. ... Does anybody have more infos on this module? Is it new or only extended? Is it available without buying the SDK? regards, klaus

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Re: [Serusers] Support for Gateway still on RFC2543
by Klaus Darilion 19 Jan '05

19 Jan '05

Simon Miles wrote: > Klaus, > > Thanks for the feedback, but I still think it is a problem. > > If I use the prefix command, this effects the URI field but not the To > field. According to RFC2543 the To field is the one to use for dialling > when the INVITE gets to it's final destination. Are you sure that the To: field is used fpr dialing - not the request URI? Can you point me to the relevant sections in RFC2543? regards, klaus PS: Please CC to the list. > > Hence the prefix command can't be used ! ! ! If I mangle the To field > then this effects the Call-ID so the SIP software sees a reply to the > INVITE as another message ! ! > > Simon > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: 10 January 2005 22:27 > To: Simon Miles > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Support for Gateway still on RFC2543 > > > There should be no problem at all - RFC 3261 is compatible with the old > RFC. ser will look for the "lr" parameter in the via headers and will > use strict routing if the lr parameter is not found in the topmost via > header. > > regards, > klaus > > Simon Miles wrote: > > >>Dear Community, >> >>I still have gateways that confirm to RFC2543 and not the newer >>RFC3261. This means the use of URI and To fields are different. Is >>there any way of telling sip_router that it needs to conform to the >>old spec ? >> >>Thanks >> >> >>Simon >> >> >> >>---------------------------------------------------------------------- >>-- >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] IPv6, problems with link local
by Michelle Mendel 19 Jan '05

19 Jan '05

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[Serusers] Regisater request with Recorde Route Header........
by innovation.interops＠wipro.com 19 Jan '05

19 Jan '05

Hello, As per RFC 3261 the Registrars must ignore Record-Route field in any incoming REGISTER request and should not include the same in the response. But I find SER implementations including the Record Route Field in the 200OK Response. Any advice or suggestion?? Thanks karthikeyan Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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[Serusers] Processing REGISTER REQ section 10.3
by innovation.interops＠wipro.com 19 Jan '05

19 Jan '05

Hello, As per section 10.3 step 6 the bindings must not be removed if there is an incoming REGISTER request with... CallID==One stored CSeq==Same as stored Contact : * Expires: 0 But I found SER Registrar implementations removing the bindings and sends 200Ok.It should have dropped the request. Any thoughts or inputs?? Thanks karthikeyan Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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[Serusers] AVP ops module question.
by Ricardo Martinez 19 Jan '05

19 Jan '05

Hello. I have a few questions about the usage of the AVP ops module for Dynamic Routes. First. Suppose that i have a table called "Prefix". This table looks like : uuid code Value 107 0056 192.168.0.10 107 0044 192.168.0.20 107 001 192.168.0.30 and maybe a few more lines like these. My first question is Does the avp_db_load( "i:107" ) command load all the "code" and "value" rows? Second : if i succesfully loaded all the rows i need to compare the RURI from the INVITE with the avp parameter. In the avp_check("i:107", "eq/$ruri/i ") i need to compare the ruri parameter with the loaded code. So if i loaded all the codes, the avp_check compare the ruri with all the avp's loaded?. Does the "eq/" check a exact match or can i use it for the "best" match?. For example 005621234567 is matched with 0056. I really hope that some can help me here. Thanks in advance Ricardo Martinez Ogalde.-

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[Serusers] SER Radius Web FrontEnd
by Michele "O-Zone" Pinassi 19 Jan '05

19 Jan '05

Hi all, i'm start working to a Web (PHP+MySQL) frontend to manage users, view calls and accounting, in SER with Radius Auth and Acc. Before loosing time, there's something already done ? Thanks ! Oz -- ---- O-Zone ! No(C) since 1996 www.zerozone.it

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[Serusers] routing question - what is "current URI" in these cases?
by Gavin Bensom 19 Jan '05

19 Jan '05

Hi, I'd like some clarification on script routing. To quote the SER Admin guide. . . " It is important to realize that ser operates over current URI all the time. If an original URI is rewritten by a new one, the original will will be forgotten and the new one will be used in any further processing. In particular, the uri matching operand and the user location action lookup always take current URI as input, regardless what the original URI was. " Question 1 - Current URI = sip:user@domain.com. What is the "current URI" after these two lines are processed in the script? What is the other uri in the destination set? rewritehostport("10.10.10.40:5095"); append_branch(); Question 2 - Current URI = sip:user@domain.com. What is the "current URI" after the same two lines are processed in the script but in the opposite order? What is the other uri in the destination set? append_branch(); rewritehostport("10.10.10.40:5095"); Question 3 - Current URI = sip:user@domain.com. What is the "current URI" after these three lines are processed in the script? Assume that user(a)domain.com has two locations. sip:user@10.10.10.20 - priority = 0.0 sip:<user_cell_phone>@<gateway_ip> - priority = 1.0 rewritehostport("10.10.10.40:5095"); append_branch(); lookup("location"); What are the other uri's in the destination set in this case? I'm confused and I've tried to determine the behavior by testing with different script configurations. The reason I'm asking is that I want my users to be able to put their cell phone numbers in with the form of sip:<cell_number>@mydomain.com Currently this does not work. To achieve proper cell phone forwarding the contact location must be: sip:<cell_number>@gateway_ip_address In previous script configurations the sip:cell_number@mydomain.com format worked. When I added these lines: rewritehostport("10.10.10.40:5095"); append_branch(); to my script before calling lookup("location"); Then SER started forwarding sip:cell_number@mydomain.com to my DNS server. :) Not what I wanted, obviously. My DNS server does have SRV records for SIP, but I'm not sure if I need to add a line in ser.cfg to make ser use SRV records. Any advice or enlightenment is appreciated. Thanks, G. __________________________________ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com

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[Serusers] click-to-dial problems
by Gavin Bensom 19 Jan '05

19 Jan '05

Hi, I'm having a very hard time getting the serweb click to dial functionality working. When I log into serweb and click on a sip address in the phonebook page, the ctd sub-window pops up and does not display any text. After about 20-30 seconds, the window displays "408 Request Timeout" in red text. This is all that happens. I cannot see any sip INVITE messages flowing via ngrep on the server. I get a entry in /var/log/httpd/access_log that shows that ctd.php was called properly. See line from access_log below. 10.10.30.113 - - [18/Jan/2005:17:53:23 -0800] "GET /user/ctd.php?target=sip:rk@mydomain.com&uri=sip:gb@mydomain.com&kvrk=1106 099602000 HTTP/1.1" 200 1323 "http://voice/user/phonebook.php?kvrk=41eda110e9af5" "Mozilla/5.0 (Windows; U; Windows NT 5.1; en-US; rv:1.7.5 ) Gecko/20041107 Firefox/1.0" I've even tried adding an echo line into the php code in the "write2fifo" function. The echoed text shows up in the ctd window when the "408 Request Timeout" message appears. I've tried the ctd.sh shell script as well. It fails to complete and I must use ctrl-c to escape it and then kill it manually. My php variable[ $this->fifo_server="/tmp/ser_fifo"; ] matches my ser.cfg script. # script excerpt check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=5 fifo="/tmp/ser_fifo" #sip_warning=no group="serfifo" fifo_mode=0666 Any ideas? I can't find any other log that shows any activity when I attempt to use click-to-dial. I really want to get this working. Thanks, G. __________________________________ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com

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[Serusers] More progress on my media proxy issue
by Doug Eubanks 19 Jan '05

19 Jan '05

Ok, I have gotten my media proxy to work fine between sip to sip calls, but I seem to have problems when I am forwarding a call to the PSTN. I am sure I just have something in the wrong order, any pointers? Earlier in the ser.cfg if (method=="INVITE" || method=="ACK") { log(1, "[SER]: Loose Route detected, using MediaProxy\n$ use_media_proxy(); }; Forwarding section..... rewritehostport("PSTNGATEWAY:5060"); forward(uri:host, uri:port); break; The speed of time is one-second per second.

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RE: [Serusers] multiple SIPURA behind same NAT
by Richard 19 Jan '05

19 Jan '05

What firewall do you have? Your firewall probably has problems to support hairpin traffic. Richard _____ From: Raymond Chen [mailto:rchen@cityabove.net] Sent: Tuesday, January 18, 2005 4:56 PM To: 'Richard' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT We use SIPURA 2000 and rtpproxy. I have attached the sip message log. Ray _____ From: Richard [mailto:richard@o-matrix.org] Sent: Wednesday, January 19, 2005 10:27 AM To: 'Raymond Chen' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT What nat device do you have? Does it have any sip helper module? Are you running rtpproxy? It will be helpful if you can send a sip packet dump. Richard _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Raymond Chen Sent: Tuesday, January 18, 2005 2:59 PM To: 'Matt Schulte' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT Matt, Both port use different sip port, and can make calls at the same time. But still can't get voice when call to each other. Raymond _____ From: Matt Schulte [mailto:mschulte@netlogic.net] Sent: Tuesday, January 18, 2005 9:08 AM To: Raymond Chen; serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT If I had a guess I would say either the NAT or rtp/media proxy is getting confused about which client is which. Ignoring the problem for a second, can both sipura's make calls at the same time? Are you using NAThelper for this with SER? You mentioned "without proxy", this would mean port forwarding? Try this, make each sipura use a diff range of RTP ports (in their config). Matt -----Original Message----- From: Raymond Chen [mailto:rchen@cityabove.net] Sent: Monday, January 17, 2005 12:25 PM To: serusers(a)lists.iptel.org Subject: [Serusers] multiple SIPURA behind same NAT Dear all, I have two sipura box behind NAT. both use via header and can dial out and received calls without proxy. but they have no voice when they call to each other, any idea? Raymond

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[Serusers] Manipulate malformed contact header *before* REGISTER processing
by Kanakatti Mahesh Subramanya 19 Jan '05

19 Jan '05

Is there any way to manipulate a malformed contact header in a REGISTER packet *before* we run it through proxy_auth, etc.? I've got a couple of phones behind a Cisco router, which insists on lopping off part of my Contact header. Basically, my inbound REGISTER packets shows the contact header as Contact: <sip:foo.bar@192.168.1.1:5060;transport=udp;line=8hzddhav>;q= (note the trailing 'q=') What i'd like to do is append a "1.0" to the header, and *then* run it through the proxy_auth, etc., etc. If I do a subst(), it doesnt seem to help. Any ideas?

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[Serusers] sems & nat
by Giovanni Balasso 19 Jan '05

19 Jan '05

hi {ser+sems}users, i've installed sems and ser with nathelper and rtpproxy all from cvs, ser and rtpproxy work fine, i can talk to people inside and outside nat, but I can hear sems ivr only if i call from inside the nat where ser+sems is runnig. Has anyone some experience with nat and sems? May I run ser+sems from inside a nat with port forwarding (as I am doing now) or will this preclude sems from working in any case? I thought I should have to pass thru rtpproxy, but I don't know how to deal with ser+sems fifos and rtpproxy. reading from the following debug line it seems that no address translation occurs in the communication between caller (192.168.1.111) and sems (192.168.1.125, the two private IPs belonging to different networks) (997) DEBUG: run (AmIcmpWatcher.cpp:133): ICMP from 192.168.1.125: type='Destination Unreachable' src=192.168.1.125:36032 dst=192.168.1.111:8000 Thanks -- Giovanni Balasso giaso(a)yahoo.it

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[Serusers] ALIAS, PLEASE HELP MEEEE!!!
by rosita sita 19 Jan '05

19 Jan '05

DEAR ALL, i'm using ser 8.12 version in redhat 9.0. i have problem when i tried to add alias. the problem are like this: when i don't add script in ser.cfg "modparam("usrloc", "use_domain", 1)", i tried to add alias: # serctl alias add 987654 sip:klien1@sip.pcr.ac.id the output said: 200 added to table alias. and the problem is I can't contact the user by alias. 404 not found when i added script in ser.cfg "modparam("usrloc", "use_domain", 1)", i tried to add alias, the ouput said: [root@gateway root]# serctl alias add 987654 sip:klien1@sip.pcr.ac.id 500 Error while adding contact ('987654(a)sip.pcr.ac.id','sip:klien1@sip.pcr.ac.id') in table 'aliases'. what should i do??? ANY IDEA????. I used the latest sc version. Sita Rosita ___________________________________________________ Share diaries and photos with your friends and family. http://www.homemaster.net - Homemaster. Come Together. Online.

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[Serusers] Bug with allow_trusted ?
by Francesco Bottà 19 Jan '05

19 Jan '05

Hi all, maybe it's a bug in module permissions. I'm trying to use the allow_trusted function to trust my SIP/DSS-1 gateways. Below my conf regarding module permission: modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") [skip] if (search("(User-Agent): Cisco-SIPGateway/IOS-12.x") && (allow_trusted())) { setflag(2); append_hf("P-hint:Trusted_GW\r\n"); route(2); break; }; but SER crashed and below the backtrace of the coredump: #0 0x0026fc32 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002ae989 in raise () from /lib/tls/libc.so.6 #2 0x002b0342 in abort () from /lib/tls/libc.so.6 #3 0x08083039 in qm_debug_frag (qm=0xbd4fb000, f=0xbd6b2e98) at mem/q_malloc.c:156 #4 0x08083425 in qm_malloc (qm=0xbd4fb000, size=20, file=0xbf0dcf "hash.c", func=0xbf0e1c "hash_table_insert", line=98) at mem/q_malloc.c:379 #5 0x00beb8b3 in _shm_malloc (size=20, file=0xbf0dcf "hash.c", function=0xbf0e1c "hash_table_insert", line=98) at shm_mem.h:135 #6 0x00beb04d in hash_table_insert (hash_table=0xbd6b2b80, src_ip=0xa1b6c58 "X.X.X.X", proto=0xa1b6c66 "any", pattern=0xa1b6c6a "sip:[0-9]*@X.X.X.X") at hash.c:98 #7 0x00befb4a in reload_trusted_table () at trusted.c:400 #8 0x00beefd4 in init_trusted () at trusted.c:129 #9 0x00bedd33 in mod_init () at permissions.c:459 #10 0x08075260 in init_mod (m=0x8140ce4) at sr_module.c:474 #11 0x080751d8 in init_mod (m=0x8140dd4) at sr_module.c:471 #12 0x080751d8 in init_mod (m=0x8140ec4) at sr_module.c:471 #13 0x080751d8 in init_mod (m=0x8140fb4) at sr_module.c:471 #14 0x080751d8 in init_mod (m=0x81410a4) at sr_module.c:471 #15 0x080751d8 in init_mod (m=0x8141194) at sr_module.c:471 #16 0x080751d8 in init_mod (m=0x8141284) at sr_module.c:471 #17 0x080752f1 in init_modules () at sr_module.c:497 #18 0x0805d197 in main (argc=2, argv=0xbff219d4) at main.c:1553 and here other trace in debug mode: 0(0) Number of rows in trusted table: 2 0(0) qm_malloc(0xbd559000, 20) called from hash.c: hash_table_insert(98) 0(0) qm_malloc(0xbd559000, 20) returns address 0xbd710de4 frag. 0xbd710dcc (size=20) on 1 -th hit 0(0) qm_malloc(0xbd559000, 13) called from hash.c: hash_table_insert(105) 0(0) qm_malloc(0xbd559000, 16) returns address 0xbd710e28 frag. 0xbd710e10 (size=16) on 1 -th hit 0(0) qm_malloc(0xbd559000, 24) called from hash.c: hash_table_insert(129) 0(0) qm_malloc(0xbd559000, 24) returns address 0xbd710e68 frag. 0xbd710e50 (size=24) on 1 -th hit 0(0) Tuple <Y.Y.Y.Y, any, sip:[0-9]*@Y.Y.Y.Y> inserted into trusted hash table 0(0) qm_malloc(0xbd559000, 20) called from hash.c: hash_table_insert(98) 0(0) BUG: qm_*: prev. fragm. tail overwritten(c0c0c000, abcdefed)[0xbd710e98:0xbd710eb0]! 0(0) qm_status (0xbd559000): 0(0) heap size= 33340380 0(0) used= 1586532, used+overhead=1801928, free=31538452 0(0) max used (+overhead)= 1801928 0(0) dumping all alloc'ed. fragments: If so, what I'm wrong with the conf? Any suggestions?? Many thanx Verbal

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[Serusers] MediaProxy Not Showing The Traffic
by Doug Eubanks 19 Jan '05

19 Jan '05

I am making good progress, however I always seem to have NAT issues with nathelper, so I am trying to use MediaProxy. I have it installed, and the calls are showing up on the Media Sessions page, but they always show as "Status: inactive" and never show any actual traffic. Also, the client that is behind NAT (or not) can not pass any voice traffic in or out. If I run the rtpgenerator.py it shows the test rtp with traffic flowing. Any ideas? Thanks, Doug

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[Serusers] Duplicate entries in the location table
by Nicky 19 Jan '05

19 Jan '05

Dear all, I have having a problem with the location table which I have duplicate entries from the same username but from different port. How can I prevent this from happening? Please help......Thank you in advance... Regards, Nicky

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[Serusers] provider registration
by tito 19 Jan '05

19 Jan '05

Hello. I have been looking for the way to register ser onto the provider sip server but I don't get the wat to. Is it possible ser registering on the provider server to redirect incoming calls to it? How can I do it? Thanks! regards

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[Serusers] ser on linksys router
by Richard 19 Jan '05

19 Jan '05

Hi, Here is a very interesting project which allows ser to run on Linksys wrt54g. http://sipath.sourceforge.net/ Richard

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[Serusers] SER-0.9.0 Multiple-domain
by Helge Waastad 19 Jan '05

19 Jan '05

Hi, I've got a question which may or may not have been answered earlier. I'm running ser-0.9.0 with two domains, <domain-a> and <domain-b>. (serctl domain add <domain-a>/<domain-b>) I've added <user-a>@<domain-a> with password <password-a> (serctl add <user-a>@<domain-a> <password-a> <email> Both modules registrar and userloc has use_domain=1 Default SIP_DOMAIN="<domain-a>" Registration with <user-a>@<domain-a> and password <password-a> works, as usual, fine. If I modify my user to register as: <user-a>@<domain-b> and password <password-a> and it still is registers OK! Is this how it should work? I would expect that the client should not be able to register using another realm... Best regards, Helge Waastad

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[Serusers] avpops module
by Richard 19 Jan '05

19 Jan '05

Hi Elena, I have a few questions in regards to avpops. Hope that you can help. 1. When avp_pushto() is used, the ruri is only used when t_relay is called. However the uri match can't match the regular expression correctly. For example, I have this, avp_write("sip:nobody@test.com", "s:default"); avp_pushto("$ruri", "s:default"); if (uri =~ "^sip:nobody@") { xlog("L_INFO", "got nobody\n"); } else { xlog("L_INFO", "didn't get nobody\n") } It would say "didn't get nobody\n". If I run t_relay, then it is sent to sip:nobody@test.com. 2. Is it possible to have a function to copy an avp to another avp? The application is this, tm module uses one avp to define the invite timer. If there are multiple timer values defined for different scenarios, we need to copy the value into the one used in tm module. For example, s:pstn_invite=100, s:sip_invite=20, If (the call is to pstn) { avp_copy("s:pstn_invite", "s:callee_fr_inv_timer"); } else { avp_copy("s:sip_invite", "s:callee_fr_inv_timer"); } 3. For performance reasons, we'd like to group all user preference in one domain in one mysql table. Is it possible to specify the table name with an avp? Regards, Richard

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[Serusers] help me
by sky0912＠vip.sina.com 19 Jan '05

19 Jan '05

hello jiri,every one: Thanks help me.i use ser-0.9.0 version ser,client is windows messenger,messager registrar to sip server,it problem. [root@localhost ser-0.9.0]# 17(25914) handle_subscription() entered 17(25914) parse_hfs(): Error while parsing headers 17(25914) handle_subscription(): Error while parsing message header 17(25914) handle_subscription about to send_reply and return -2 17(25914) SUBSCRIBE execute 17(25914) handle_subscription() entered 17(25914) parse_hfs(): Error while parsing headers 17(25914) handle_subscription(): Error while parsing message header 17(25914) handle_subscription about to send_reply and return -2 17(25914) SUBSCRIBE execute 13(25907) handle_subscription() entered 13(25907) parse_hfs(): Error while parsing headers 13(25907) handle_subscription(): Error while parsing message header 13(25907) handle_subscription about to send_reply and return -2 13(25907) SUBSCRIBE execute 18(25915) Binding '1008(a)10.167.16.179','sip:10.167.16.54:10749' has expired 17(25914) handle_subscription() entered 17(25914) parse_hfs(): Error while parsing headers 17(25914) handle_subscription(): Error while parsing message header 17(25914) handle_subscription about to send_reply and return -2 17(25914) SUBSCRIBE execute 12(25906) handle_subscription() entered 12(25906) parse_hfs(): Error while parsing headers 12(25906) handle_subscription(): Error while parsing message header 12(25906) handle_subscription about to send_reply and return -2 12(25906) SUBSCRIBE execute This is my scripts # ------------------ 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/textops.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) # 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_db", "db_url","mysql://ser:heslo@10.167.16.179/ser") modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("pa", "pa_domain", "10.167.16.179") modparam("pa", "db_url", "mysql:ser:heslo@10.167.16.179/ser") modparam("pa", "default_expires",50) #modparam("pa", "default_expires",3600) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # if (src_ip==193.175.135.0/24){ # force_send_socket(smaug:5080); # forward(193.175.135.179); # break; # } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="SUBSCRIBE") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); log(1,"SUBSCRIBE execute\n"); break; }; # if (method=="PUBLISH") # { # if (method=="PUBLISH") # { # if (!t_newtran()) # { # log(1, "newtran error\n"); # sl_reply_error(); # }; # handle_publish("registrar"); # log(1,"PUBLISH execute\n"); # break; # }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ______________________________________ ===================================================================

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[Serusers] can not confirm
by Pham Minh Dung 19 Jan '05

19 Jan '05

I installed SER, SER-MYSQL I installed Apache, PHP, and SERWEB. The database for ser is good. When I use the form to subscribe a user, I receive a link to confirm. When I confirm, the database has the user that I register (I check the database). But the confirmation.php can not open, and I can not login by the username, password that I register. So, help me, please. Sorry for my english. Thanks. -------------------------------------------------------- Pham Minh Dung Vietnam New Generation Telecom Adress: 4A2 Nguyen Chi Thanh - Hanoi Tel: 7753014 - 7712777

1 0

[Serusers] User interface website
by Tracy Chan 19 Jan '05

19 Jan '05

Hi All, Could anyone tell me that what the usage and purpose of 'send im', 'message store' and 'voicemail' functions which are all in user interface website ? Thank you very much. Regards, Tracy

2 1

[Serusers] ACL
by Tracy Chan 19 Jan '05

19 Jan '05

Hi All, Do anyone know what Access-Control-List (ACL) is ? Thank you very much. Regards, Tracy

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