- sr-users - kamailio.org

Re: [Serusers] call forward no answer (solved)
by caio 25 Aug '08

25 Aug '08

append_branch() is missed at ser examples... On Fri, Aug 15, 2008 at 4:22 PM, caio <elcaio(a)gmail.com> wrote: > Hi all, > I can't make call forwarding on no answer works on my ser-0.9.7 installation. > Here you can see a snippet of log, where the call falls on > failure_route and then to route_1 where t_relay is called. > > Log at: http://rafb.net/p/xwTmn195.html > > The call is from sip:579951 to sip:579915(who has enabled the feature > fwdnoanswer to sip:579916. > But I do not know why the INVITE after the append_branch() is still to > R-URI: sip:579915@.... > > After the relay then the call terminates with: > > Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ > Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ > Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ > Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ > > I follow the example of ser guide.., but do not work.. > I would appreciate any comment.. > > Regards, > > -- > caiogf > -- caio

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[Serusers] ser + media server
by caio 25 Aug '08

25 Aug '08

Hi all, What do you recommend for media server application to integrate wih ser-0.9.x? SEM or Asterisk? Specifically i'm talking about running services such as IVR, conference and voicemail.. Thanks your suggestions.. -- caio

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[Kamailio-Users] crashing problem
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

We're using Carrier route with about 53,000 routes stored in a mysql databse. We're getting this crash in production that we never had in the lab. The bt is below. Does anyone have any suggestions? nline static void free_to_params(struct to_body* tb) { struct to_param *tp=tb->param_lst; struct to_param *foo; while (tp){ foo = tp->next; <-- **** Line 75 ***** pkg_free(tp); tp=foo; } } -Jonathan #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 (gdb) backtrace full #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 No locals. #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 __FUNCTION__ = "clean_hdr_field" #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 No locals. #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 No locals. #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 msg = (struct sip_msg *) 0x8193030 __FUNCTION__ = "receive_msg" #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 len = 1271 buf = "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"... tmp = <value optimized out> from = (union sockaddr_union *) 0x8a91318 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {2850040017, 135426432, 3213031528, 3085483479}, addr32 = {2850040017, 135426432, 3213031528, 3085483479}, addr16 = {10449, 43488, 29056, 2066, 63592, 49026, 48599, 47080}, addr = "Ñ(à©\200q\022\bhø\202¿×½è·"}}, dst_ip = {af = 2, len = 4, u = {addrl = {509300942, 0, 0, 0}, addr32 = {509300942, 0, 0, 0}, addr16 = {20686, 7771, 0, 0, 0, 0, 0, 0}, addr = "ÎP[\036", '\0' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto = 1, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023ÄÑ(à©\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 2850040017}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 2850040017, sin6_addr = {in6_u = {u6_addr8 = '\0' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x818dd90} __FUNCTION__ = "udp_rcv_loop" #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 cfg_log_stderr = 0 cfg_stream = (FILE *) 0xa179008 c = <value optimized out> r = <value optimized out> tmp = 0x8148b9c "ÿÿÿÿ" tmp_len = <value optimized out> port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 639236262 rfd = 4 __FUNCTION__ = "main"

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Re: [Kamailio-Users] Google reference to openser users and devel Archives
by H Quintana 25 Aug '08

25 Aug '08

Hi, My temporary solution was to edit the etc/hosts file in my windows xp machine. The path is: C:\WINDOWS\system32\drivers\etc\hosts And I added the lines: 193.22.119.66 www.openser.org 193.22.119.66 openser.org to point to the Kamailio servers. Hope this helps, Humberto > Subject: Re: [Kamailio-Users] Google reference to openser > users and > devel Archives > To: users(a)lists.kamailio.org > Message-ID: > <200808251326.44077.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Friday 15 August 2008, Rafael Risco Gonzales-Vigil > wrote: > > This is for openser/kamailio and opensips list > administrators: > > google still has references to > http://www.openser.org/pipermail/users/ and > > http://www.openser.org/pipermail/devel/ for openser > related searchings and > > those urls does not contains old files anymore so its > dificult search > > information for learning and troubleshooting about > openser/kamailio or > > opensips... I hope someone can solve this issue for > the benefit of all > > current and future community users. > > Hi Rafael, > > we don't have access to the openser.org domain, so > there is nothing what we > can do here, sorry. All access of web and mail services to > openser.org will > go to opensips infrastructure. > > If you encounter such old links with openser.org, you can > simply > change "openser.org" to "kamailio.org", > then you'll be able to access the > content, apart from using the google archives, as Alex > pointed out. > > Cheers, > > Henning >

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[Kamailio-Users] failure route configuration
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

I have this block in my configuration: <excerpt from failure_route[1]> if(!cr_rewrite_uri("1", "call_id")){ t_reply("403", "$fU@$fd Not allowed"); } else { # In case of failure, re-route the request prefix("+"); t_on_failure("2"); if (is_method("INVITE")) t_on_reply("1"); # Relay the request to the gateway t_relay(); } The output in my log file: Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5622]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_one Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: Failure Route Started Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_two Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:t_forward_nonack: no branch for forwarding Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:w_t_relay: t_forward_nonack failed What am I missing that I need to do prior to calling t_relay()? -Jonathan

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[Kamailio-Users] Google reference to openser users and devel Archives
by Rafael Risco Gonzales-Vigil 25 Aug '08

25 Aug '08

This is for openser/kamailio and opensips list administrators: google still has references to http://www.openser.org/pipermail/users/ and http://www.openser.org/pipermail/devel/ for openser related searchings and those urls does not contains old files anymore so its dificult search information for learning and troubleshooting about openser/kamailio or opensips... I hope someone can solve this issue for the benefit of all current and future community users. rafael Lima - Peru

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[Kamailio-Users] need help on starting opensips v1.4.1
by Do Nguyen Ha 25 Aug '08

25 Aug '08

hi opensips debian:/etc/opensips# /etc/init.d/opensips start Starting opensips: opensipsListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.254 [192.168.1.254]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.254 [192.168.1.254]:5060 Aliases: tcp: debian:5060 tcp: localhost:5060 udp: debian:5060 udp: localhost:5060 . debian:/etc/opensips# /etc/init.d/opensips status Status of opensips: opensips is not running. could anyone tell me the tips for this Thanks Ha`

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[Kamailio-Users] Why does Kamailio reply 408 when 'fr_inv_timer' expires?
by Iñaki Baz Castillo 25 Aug '08

25 Aug '08

Hi, when 'fr_inv_timer' expires Kamailio CANCELs the pending branches and reply "408 Request Timeout" to the UAC. Well, I wonder if 408 is appropiate. I think "480 User Unavailable" or "487 Cancelled" would be better: I received those replies in other maillist: "408 indicates transaction timeout. You have sent request to next hop and it has not responded back. If the gateway has not responded at all (Including 100 trying), 408 is appropriate. 480 is more of application usage meaning that the user is not available and would be appropriate if the gateway has at least send one provisional response i.e. 100, 180, 183..." "408 is used consistently to mean IP-level timeout in sending a request from one element to another. It is also defined to be returned by a UA when the time specified in "Expires" has been reached. (In theory, two different error codes should be defined for these two situations, but that has not been done.) However, in a ring-no-answer situation, it is more common for an upstream proxy to send a CANCEL to the UA/gateway, in which case the UA/gateway provides a 487 response to the INVITE." This last point suggest that Kamailio could do the following: - If the proxy has received provisional responses from callee's then not send a 408 upstream since when the proxy sends CANCEL to UAS's it will receive some 487 and would return a 487 upstream. - If the proxy hasn't received provisional responses then this is a clear "408 Request timeout". So a possibility is that Kamailio just reply a 408 in real timeouts and reply 487 when 'fr_inv_timer' expires (or just reply nothing since UAS's will generate the 487). Opinions? -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Where have the OLD Module Docs gone?
by Douglas Garstang 25 Aug '08

25 Aug '08

The module documentation for versions 1.2, 1.3 etc used to be available on the web site. Where has it gone? When you clicked on the documentation link, you used to be able to choose the version of OpenSER(Kam...OpenSIPS whatever) that you wanted. Now, your taken immediately to what appears to be the documentation for the LATEST version. Doug.

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[Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

Hi, lists I have some problems with the simple presence in kamailio 1.4, I add to my kamailio.cfg the following line and when restarting the it shows me errors in this it lines Aug 24 19:31:41 [21946] ERROR:core:set_mod_param_regex: parameter <force_active> not found in module <presence> Aug 24 19:31:41 [21946] CRITICAL:core:yyerror: parse error in config file, line 50, column 19-20: Can't set module parameter Aug 24 19:31:41 [21946] ERROR:core:main: bad config file (1 errors) Aug 24 19:31:41 [21946] NOTICE:presence:destroy: destroy module ... I remove of the Makefile the modul presence kamailio.cfg loadmodule "presence.so" modparam("presence|auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:192.168.10.1:5060") where the error can be? regards rickygm

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[Kamailio-Users] preparing openser 1.3.3
by Daniel-Constantin Mierla 24 Aug '08

24 Aug '08

Hello, I propose to build a new minor release for openser 1.3 branch on the 27th of August (Wednesday). There are already a lot of fixes since 1.3.2. Probably couple of backports from 1.4 will get in until Wednesday, the rest, if it is the case, shall get in 1.3.4, probably in about another month, as we are going to maintain 2 stable branches. Perhaps 1.4.1 will be out by same time or so. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] FreeRadius patch for OpenSer1.2.2
by Nuno Marques 24 Aug '08

24 Aug '08

Hi, I'm following the examples on the book "Building Telephony Systems with OpenSer". I came to Accounting and Billing chapter using FreeRadius. Flavio (the author) says we need to apply a patch to the freeradius to "fix a problem with accounting type 15". My question is - is it still necessary to apply that patch or the new version of freeradius allready covers it? I'm having big problems applying that patch because of the key signature problems. Appreciate any help in this subject. Thanks in advance, Nuno

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[Kamailio-Users] AUTO: Miodrag Radulovic is out of the office (returning 07-09-2008)
by Miodrag Radulovic 23 Aug '08

23 Aug '08

I am out of the office until 07-09-2008. Privremeno nedostupan - Out-of-Office Message. Poruka koju ste poslali je prispela i bice sacuvana, ali napominjem da sam na odmoru van zemlje do 07.09.2008. i moj pristup elektronskoj posti bice veoma ogranicen. Ukoliko se radi o potrebi za hitnom intervencijom, dostupan sam preko mobilnog telefona 065/3016576. The message that you have sent will be saved in my inbox, but please note that due to summer holliday I will be out of the country until 07.09.2008. In that period my access to my email will be very limited. In case of urgency, I am available on my mobile phone: +381 65 3016576. Miodrag Radulovic General Manager COMUTEL Note: This is an automated response to your message "Users Digest, Vol 39, Issue 83" sent on 23.8.2008 0:49:18. This is the only notification you will receive while this person is away.

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Re: [Serusers] SER - Waiting Transaction
by Arun Kumar 22 Aug '08

22 Aug '08

Hi, thanks for your reply. I'll ignore this but I see so many open connection to MySQL from SER and they don't get close automatically also resulting in denial of other service using my MySQL. Thanks arun On Mon, Aug 18, 2008 at 10:12 PM, Jiri Kuthan <jiri(a)iptel.org> wrote: > ignore it > > Arun Kumar wrote: > >> Hi, >> >> >> I'm running SER 0.96 in my production for a long but there is strange >> problem , when I do serctl moni and looks for transaction it shows waiting >> transaction in () almost 400+ can some one tell me why there are so many >> transaction in waiting while I see processing only 10-20 transaction. >> >> Thanks >> arun >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >

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[Kamailio-Users] Problem to run MediaProxy 2.0.3
by Leonardo Santana 22 Aug '08

22 Aug '08

Hi all, I´m using mediaproxy 2.0.3 on OpenSuSE 11.0. When I run the "media-dispatcher" I see this message bellow: PS.: The OpenSuSE 11.0 doesn´t have the module "libgnutls-extra.so.13" only the libgnutls-extra.so.26". ######## Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Log opened. Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Starting MediaProxy Dispatcher 2.0.3 Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Traceback (most recent call last): Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/media-dispatcher", line 46, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.dispatcher import Dispatcher Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/dispatcher.py", line 31, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.tls import X509Credentials, X509NameValidator Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/tls.py", line 12, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls import crypto Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/crypto.py", line 11, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.validators import method_args, none, ignore, list_of, one_of Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/validators.py", line 9, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.constants import * Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/constants.py", line 38, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import constants Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/__init__.py", line 7, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import errors Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/errors.py", line 14, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library.functions import gnutls_strerror, gnutls_alert_get Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/functions.py", line 6, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] _libraries['libgnutls-extra.so.13'] = CDLL('libgnutls-extra.so.13') Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/ctypes/__init__.py", line 348, in __init__ Aug 21 17:18:32 nana media-dispatcher[5502]: [-] self._handle = _dlopen(self._name, mode) Aug 21 17:18:32 nana media-dispatcher[5502]: [-] OSError: libgnutls-extra.so.13: cannot open shared object file: No such file or directory ##### Thanks a lot!!!

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[Kamailio-Users] RFC: branches/location records control
by Daniel-Constantin Mierla 22 Aug '08

22 Aug '08

Hello, one of the most hidden parts of openser to the config is the branch processing. Behind a branch is following structure: struct branch { char uri[MAX_URI_SIZE]; unsigned int len; /* Real destination of the request */ char dst_uri[MAX_URI_SIZE]; unsigned int dst_uri_len; /* Path vector of the request */ char path[MAX_PATH_SIZE]; unsigned int path_len; int q; /* Preference of the contact among contact within the array */ struct socket_info* force_send_socket; unsigned int flags; }; I am thinking in offering a fine access to this structure in configuration file, so the admin can control much closer the branch processing. By using while and and index, checks of the attributes in branches could be done. By doing it in main route, one can reduce the overload of stateful processing, by dropping a branch before going to transaction, for example. I am looking at something similar for user location records. Such a record has many attributes, and from config is no way to access them without doing a lookup and changing r-uri/branches. There could be many benefits, for example: - being able to detect if a user calls from a registered phone or not - detecting how many contacts a user has at a moment - ability to check and, eventually, drop processing before doing changes to branches and so on... The access will be provided via pseudo-variables, perhaps the location records will be fetched in variables upon calling a function that gets an AoR as parameter. I have seen on mailing lists discussions about needs that will be solved by this new feature. Any comments? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] kamailio-1.4.0 and SerMyAdmin
by luzango mfupe 22 Aug '08

22 Aug '08

Hi, Im bit cnfsd, does SerMyAdmin support Kamailio-1.4.0?. I did my configuration using SerMyAdmin 0.7-Beta.war and then 0.7-beta.war files, but everytime i pointed my browser to the IP address of my installation my Tomcat6 server return the message that SerMyAdmin does not exist!!! Can someone pls'e clarify this?? .. wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] openser 1.2.1 errors in the log
by toly 22 Aug '08

22 Aug '08

Greetings, Proxy compiled with TLS support, tls is not turned on Routing script inclulded below. Underlying database is Sybase. The following is happening: There are 25 users having as a contact each other. Softphone is Bria. Without any calls, just subscribe/notify proxy is running out of the socket in one hour. The users are or outside coming via corp firewall, or if inside, the route is setup the way the going via corp firewall, so they are natted. Proxy is in DMS on the same network as firewall. There are no problems with nat traversal. Daniel, could you please take look at the script? Thanks a lot, Toly. this is from log: <receive.c: 206> receive_msg: cleaning up 08/15 20:12:58 29156 debug <ip_addr.c: 109> tcpconn_new: new tcp connection to: xxx.xxx.xxx.xx 08/15 20:12:58 29156 debug <tcp_main.c: 402> tcpconn_new: on port 58544, type 2 08/15 20:12:58 29156 debug <tcp_main.c: 497> tcpconn_add: hashes: 96, 45 08/15 20:12:58 29156 debug <tcp_main.c: 993> handle_new_connect: new connection: 0xb6485590 98 flags: 0002 08/15 20:12:58 29156 debug <tcp_main.c: 935> WARNING: send2child: no free tcp receiver, connection passed to the least busy one (1) this is routing script: log_facility=LOG_LOCAL2 debug=7 fork=yes log_stderror=no listen=PROXY_ENV # This is where OpenSER installed port=5070 children=20 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true disable_tls = 1 listen = tls:PROXY_ENV:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = SSLv23 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" ###########################################################33 ###########################################################33 ###########################################################33 mpath = "/usr/local/lib/openser/modules" loadmodule "uac.so" loadmodule "psybase.so" loadmodule "pike.so" modparam("pike", "remove_latency", 120) modparam("pike", "reqs_density_per_unit", 10000) modparam("pike", "sampling_time_unit", 60) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0666 ) modparam("mi_fifo", "reply_dir", "/tmp/" ) modparam("mi_fifo", "reply_indent", "\t" ) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "uri.so" loadmodule "uri_db.so" modparam("uri_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") loadmodule "alias_db.so" modparam("alias_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") #modparam("dispatcher", "dst_avp", "$avp(i:271)") #modparam("dispatcher", "grp_avp", "$avp(i:272)") #modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "flags", 2 ) # failover mode set flag=2 loadmodule "usrloc.so" modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "acc.so" # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) #modparam("acc", "log_flag", 0 ) modparam("acc", "db_flag", 1 ) modparam("acc", "log_missed_flag", 1 ) modparam("acc", "db_missed_flag", 1 ) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("acc", "log_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "report_cancels", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "domain.so" modparam("domain", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "mediaproxy.so" modparam("mediaproxy","natping_interval", 10) modparam("mediaproxy","mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "received_avp", "$avp(i:801)") modparam("nathelper", "sipping_from", "sip:pinger@parusivoip.net") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 7) loadmodule "registrar.so" modparam("registrar", "received_avp", "$avp(i:801)") modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 50) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "permissions.so" modparam("permissions", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("permissions", "db_mode", 1) ######################################################################## ######################################################################## ######################################################################## route { if ( is_method("REGISTER") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON REGISTER on IP=$si !!"); exit; } }; if ( is_method("INVITE") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON INVITE on IP=$si !!"); exit; } }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; xlog("L_NOTICE", "TRACKING [0] NEW REQUEST - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); force_rport(); # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && nat_uac_test("19")) { record_route_preset("PROXY_ENV:5070;nat=yes"); # This is where OpenSER installed xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE PRESET- M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else if (method!="REGISTER") { xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); record_route(); }; append_hf("P-hint: rr-enforced\r\n"); if ( search("User-Agent: Parus.*") && is_method("BYE|REFER")) { # check the transport of callee $avp(i:44) = $(ru{uri.user}); if ( $avp(i:44) != $tU ) { $avp(i:45) = $avp(i:44); } else { $avp(i:45) = $tU; } avp_db_query("exec oserIsTcpTransport @to_user='$avp(i:45)', @td='$td'", "$avp(i:11)"); xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si AVP11=$avp(i:11) AVP44=$avp(i:44) TD=$td"); if ( $avp(i:11) == "tcp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tcp"); } else if ( $avp(i:11) == "tls" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TLS - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tls"); } else if ( $avp(i:11) == "udp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS UDP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=udp"); } else { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS NOT TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); } } # ----------------------------------------------------------------- # ACC Section # ----------------------------------------------------------------- if ( is_method("CANCEL|BYE|INVITE|MESSAGE") ) { setflag(1); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if ( is_method("BYE|CANCEL") ) { xlog("L_NOTICE", "TRACKING [0] CALL TEAR DOWN: END_MEDIA_SESSION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); if (!has_totag()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if ( is_method("INVITE") ) { if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorized - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); }; }; if ( (nat_uac_test("19") || search("^Route:.*;nat=yes")) && is_method("INVITE|REFER|ACK") ) { setbflag(6); # NONONO fix_nated_contact() xlog("L_NOTICE", "TRACKING LOOSE_ROUTE using media proxy - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); use_media_proxy(); }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if ( !is_uri_host_local() ) { if ( is_from_local() || allow_trusted() ) { xlog("L_NOTICE", "TRACKING[0] IS_FROM_LOCAL OR TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else { xlog("L_NOTICE", "TRACKING[0] NOT IS_FROM_LOCAL OR NOT TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Forbidden"); exit; } }; if ( is_method("CANCEL|BYE") ) { route(4); exit; } else if ( is_method("ACK") ) { route(1); exit; } else if ( is_method("INVITE") ) { route(3); exit; } else if ( is_method("REGISTER") ) { route(2); exit; } else if ( is_method("OPTIONS") ) { route(11); exit; } else if ( is_method("MESSAGE") ) { xlog("L_NOTICE", "TRACKING [14] ROUTE14 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); route(14); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; xlog("L_NOTICE", "TRACKING[0] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { xlog("L_NOTICE", "TRACKING [1] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); if(isbflagset(6)) { xlog("L_NOTICE", "TRACKING [1] USING NATed reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("1"); } else { xlog("L_NOTICE", "TRACKING [1] USING Standard reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("2"); } if (!t_relay()) { xlog("L_NOTICE", "TRACKING [1] t_relay() Failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; exit; } # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); setbflag(7); xlog("L_NOTICE", "TRACKING [2] Fixed nated register - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); } if (!save("location")) { xlog("L_NOTICE", "TRACKING [2] Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); sl_reply_error(); }; xlog("L_NOTICE", "TRACKING [2] Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { xlog("L_NOTICE", "TRACKING [3] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING [3] proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING [3] spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); }; if (client_nat_test("3")) { fix_nated_contact(); setbflag(6); xlog("L_NOTICE", "TRACKING [3] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; if ( alias_db_lookup("dbaliases") ) { xlog("L_NOTICE", "TRACKING [3] callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } if ( !is_domain_local("$rd") ) { xlog("L_NOTICE", "TRACKING [3] call for foreign domain M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(5); route(1); exit; } if ( does_uri_exist() ) { xlog("L_NOTICE", "TRACKING [3] callee was local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (lookup("location")) { xlog("L_NOTICE", "TRACKING [3] LOOKUP LOCATION SUCC - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); append_hf("P-hint: usrloc applied\r\n"); remove_hf("Alert-Info"); if ( proto==tls ) { append_hf("Alert-Info: 1\r\n"); } route(5); t_on_branch("1"); route(1); exit; } }; xlog("L_NOTICE", "TRACKING [3] SOFTPHONE LOOKUP LOCATION NOT SUCC M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); # # call pstn handler # if ( route(7) ) { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler successful M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # will never hit it } else { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler failed M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # exit for now } # # never hit this for now, possibly in the future if # forward to different domain or so # xlog("L_NOTICE", "TRACKING[3] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); exit; } # ----------------------------------------------------------------- # PSTN call section # ----------------------------------------------------------------- route[7] { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (uri =~ "^sip:011.*@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION ATTEMPT TO MAKE INTERNATIONAL CALL M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); }; if ($(tU{s.len}) > 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION MORE THEN 11 DIGITS M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); } if ($(tU{s.len}) == 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION STRIP ONE DIGIT M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); } if ( ($(rU{s.len}) != 10) && ($(rU{s.len}) != 2) && ($(rU{s.len}) != 3) && ($(rU{s.len}) != 4) && ($(rU{s.len}) != 1) ) { xlog("L_NOTICE", "TRACKING [37] S.LEN != 10 or 1 or 2 or 3 or 4 M=$rm RURI=$ru F=$fu T=$tu rU=$rU IP=$si ID=$ci"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [7] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; if (uri =~ "^sip:[2-9]{1}[0-9]{2}[2-9]{1}[0-9]{2}[0-9]{4}@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN OUTBOUND $fU -> $ruri "); avp_db_query("exec oserGetCallIdForBvox @fU='$fU', @from_domain='$fd'", "$avp(i:12)"); uac_replace_from("$avp(i:12)"); xlog("L_NOTICE", "TRACKING [37] REPLACED FROM PSTN avp = $avp(i:12[1]) $fU -> $ruri "); route(5); # select next gateway simulates NEXTONE # 1 - DEV, 2 - PROD asterisk, 3 - john's nextone, 4 - prod nextone, 5 - qa ds_select_domain("GWGROUPID", "4"); t_on_failure("2"); xlog("L_NOTICE", "TRACKING [37] DISPATCHER FORWARD PSTN $fu -> $ruri "); route(1); return (1); # will never hit this } else { xlog("L_NOTICE", "TRACKING [37] Attempt to dial invalid number $fu -> $tu"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [37] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; } # ----------------------------------------------------------------- # DEV / TEST section # ----------------------------------------------------------------- route[6] { xlog("L_NOTICE", "TRACKING [6] DEV / TEST SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # CANCEL or BYE handler # ----------------------------------------------------------------- route[4] { xlog("L_NOTICE", "TRACKING [4] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (client_nat_test("3")) { fix_nated_contact(); xlog("L_NOTICE", "TRACKING [4] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; route(1); } # ----------------------------------------------------------------- # NAT traversal section # ----------------------------------------------------------------- route[5] { xlog("L_NOTICE", "TRACKING [5] NAT TRAVERSAL SECTION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (isbflagset(6)) { if (!isflagset(8)) { setflag(8); xlog("L_NOTICE", "TRACKING [5] BFLAG 6 IS SET, USING MEDIA PROXY - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); avp_printf("$avp(i:450)", "$dd"); if (avp_check("$avp(i:450)", "eq/$src_ip/g")) { xlog("L_NOTICE", "TRACKING [5] LOOKUP SUCC, FLAGS 6 AND 7 BEHIND SAME NAT $fu -> $tu $dd $src_ip"); } use_media_proxy(); }; }; } # ------------------------------------------------------------------------ # OPTIONS Message Handling # ------------------------------------------------------------------------ route[11] { xlog("L_NOTICE", "TRACKING [11] OPTIONS - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } xlog("L_NOTICE", "TRACKING [11] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("200", "Got it"); exit; } # ----------------------------------------------------------------- # NAT reply route # ----------------------------------------------------------------- onreply_route[1] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] NAT REPLY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); if (nat_uac_test("1")) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] CLIENT_NAT_TEST FIXED NATED CONTACT M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); fix_nated_contact(); }; # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- $rs $rr\n" ); } if ( isbflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { if ( $cl > 0 ) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] USE MEDIA PROXY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); use_media_proxy(); }; }; exit; } # ----------------------------------------------------------------- # standard reply route # ----------------------------------------------------------------- onreply_route[2] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[2] STANDARD REPLY- M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- $rs $rr\n" ); } exit; } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[1] { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(t_check_status("408|500|503")) { if ( t_check_status("408") ) { if ( ds_next_domain() ) { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] TRYING NEXT GW - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_on_failure("1"); route(1); return; } } else { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] NOT 408 STATUS - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_reply("503", "Service not available"); return; } } } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[2] { # If fr_timer expires t_check_status("408") is true, although $rs is <null> if( t_check_status("408") ){ xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- TIMEOUT for Gateway $rd\n" ); } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- $rs reason $rr\n" ); } # basically what we do is # 1. not worry about reasonable failures (ie busy, wrong number etc...) # 2. go to next destination gateway # 3. if no destination gateway then 503 (service unavailable) # 403 -- typically ISDN network says 'not a valid number' etc.. if( t_check_status("403") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Forbidden -> ISDN Cause Code 1\n" ); end_media_session(); return; } # 408 -- timeout -- typically the end party has not answered # Since we cancel t_on_failure() on a provisional response we should not be # getting a 408 timeout from a gateway at this stage.. it will just "fall through" #if( t_check_status("408") ){ # xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Timeout\n" ); # return; #} # 486 -- User Busy if( t_check_status("486") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Destination Busy\n" ); end_media_session(); return; } # 487 -- Request Cancelled (usually in response to a CANCEL transaction) if( t_check_status("487") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Request Cancelled\n" ); end_media_session(); return; } # ok... so at this stage we try the next gateway (unless we don't have one) # if no next gateway we bail... if( ds_next_domain() ){ #t_on_reply("1"); t_on_failure("2"); xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci Next gateway $fU -> $tU via $rd\n" ); route(1); return; } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci No more gateways -> 503.\n" ); end_media_session(); t_reply("503", "Service unavailable -- no more gateways" ); return; } } # ----------------------------------------------------------------- # BRANCH route # ----------------------------------------------------------------- branch_route[1] { if (isbflagset(6)) { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } else { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as not NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } } # ------------------------------------------------------------------------ # MESSAGE Message Handling # ------------------------------------------------------------------------ route[14] { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp19007938p19007938.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] openser 1.2.1 errors in the log - tcp blocking
by toly 22 Aug '08

22 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS, TLS is not turned on. This is part of the config file: listen=xxx.xxx.xxx.xxx # This is where OpenSER installed port=5070 children=25 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true here's the problem: I have 25 users having each other as a contact. Softphone is Bria, transport is tcp here comes subscribe: 08/21 15:07:46 8029 debug <loose.c: 115> find_first_route: No Route headers found 08/21 15:07:46 8029 debug <loose.c: 883> loose_route: There is no Route HF 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [18] - M=SUBSCRIBE RURI=sip:dmarkiew@tamtam.com F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100 .15 ID=YTBmOTJhNGI5M2RmZjYyNmM2NGFhMmI1NmRjMmY4ZWE. 08/21 15:07:46 8029 debug <dset.c: 445> rewrite_uri: Rewriting Request-URI with 'sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26' 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 0 flags C0 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 1 flags C0 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-URI=sip:76.222.53.11 2:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] USING NATed reply - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-UR I=sip:76.222.53.112:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 debug <t_lookup.c: 1037> DEBUG: t_newtran: T on entrance=0xffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=ffffffffffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=78 08/21 15:07:46 8029 debug <t_lookup.c: 407> t_lookup_request: start searching: hash=41567, isACK=0 and later I get this: 08/21 15:07:48 8029 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 2 s elapsed from 2 s 08/21 15:07:48 8029 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/21 15:07:48 8029 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/21 15:07:48 8029 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/21 15:07:48 8029 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/21 15:07:48 8029 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one And I get a lot of it. Eventually I may get SO_ERROR(111) and then I have to restart proxy. I can post entire config script if anybody had/resolved/willing to take a look at the problem. Thanks a lot, Toly -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log---tcp-blocking-tp1909… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Res: Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Edson, Please check the tomcat logs. They can be found at /usr/local/tomcat6/logs. When tomcat6 don't start, usually is a typo in the configuration files. Best regards, Flavio E. Goncalves ----- Mensagem original ---- De: Edson - Listas <4lists(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: Asim Riaz <ariaz78(a)gmail.com>; users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 14:05:27 Assunto: Re: [Kamailio-Users] Res: Help with SerMyAdmin Yesterday, we faced a similar/related problem, but because Tomcat process was killed. When we try to run it again, SerMyAdmin refuse to start, altought all configurations and DB definitions where there... Any ideas on how to restore it? Edson. Flavio Goncalves escreveu: > Hi Assim, > > I'm glad it is working. SerMyAdmin creates the database fields during > the first run, that's why it worked after restarting the tomcat. The sql > file creates the required users. I'm working to have a debian package > for SerMyAdmin soon. > > Regards, > > Flavio E. Goncalves > > ----- Mensagem original ---- > De: Asim Riaz <ariaz78(a)gmail.com> > Para: users <users(a)lists.kamailio.org> > Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 > Assunto: Re: [Kamailio-Users] Help with SerMyAdmin > > Its working thanks for the help. > I removed my existing serMyAdmin directory from tomcat, restarted the > tomcat and run the script mysql -u root -p openser <openser.sql > > regards > Asim Riaz > > On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com > <mailto:yangsengopenser@gmail.com>> wrote: > > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql > script - admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com <mailto:ariaz78@gmail.com>> > > When I am adding a user i am getting > java.lang.NullPointerException; > > also I cant login using admin username. > > On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques > <yangsengopenser(a)gmail.com <mailto:yangsengopenser@gmail.com>> > wrote: > > Hi again, > > I was looking at the database schema of openser and i > noticed that the subscriber table only has 12 columns till rpid. > Looking at the openser.sql script, it has more than 12 > columns (including the 'version' column...). > What is wrong? Should SerMyAdmin add this columns to the > subscriber table? > Is it lacking me something in the database subscribet table? > Any help with this? > > Thanks in advance, > > Nuno > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com > <mailto:muhammad.akl@gmail.com>> > > > > Hi Nuno > > i faced some problems your already facing and succeeded > to solve them so i'd like to as you > > did you install SERweb tables ? > > if though the problems come from there especially the > phplib_id field a duplication happened between it and > the "id" filed so you have to remove SERweb tables and > all thing will go smoothly without any errors > > p.s i recommend you to remove the whole database e.g > openser and install it again using openser_mysql.sh script > > e.g /path/to/openser_mysql.sh create > > press yes for presence and that stuff but for SERweb no > > try and tell me what you get > > BR > > Muhammad > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Assim, I'm glad it is working. SerMyAdmin creates the database fields during the first run, that's why it worked after restarting the tomcat. The sql file creates the required users. I'm working to have a debian package for SerMyAdmin soon. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Asim Riaz <ariaz78(a)gmail.com> Para: users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 Assunto: Re: [Kamailio-Users] Help with SerMyAdmin Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: How did you add the user? With openserctl or with sermyadmin? Note that you have to use the same login/pass that are on the sql script - admin@setup for login and secret for pass. 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> When I am adding a user i am getting java.lang.NullPointerException; also I cant login using admin username. On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: Hi again, I was looking at the database schema of openser and i noticed that the subscriber table only has 12 columns till rpid. Looking at the openser.sql script, it has more than 12 columns (including the 'version' column...). What is wrong? Should SerMyAdmin add this columns to the subscriber table? Is it lacking me something in the database subscribet table? Any help with this? Thanks in advance, Nuno 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> Hi Nuno i faced some problems your already facing and succeeded to solve them so i'd like to as you did you install SERweb tables ? if though the problems come from there especially the phplib_id field a duplication happened between it and the "id" filed so you have to remove SERweb tables and all thing will go smoothly without any errors p.s i recommend you to remove the whole database e.g openser and install it again using openser_mysql.sh script e.g /path/to/openser_mysql.sh create press yes for presence and that stuff but for SERweb no try and tell me what you get BR Muhammad _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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Re: [Kamailio-Users] Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql script - > admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> > >> When I am adding a user i am getting java.lang.NullPointerException; >> >> also I cant login using admin username. >> >> On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: >> >>> Hi again, >>> >>> I was looking at the database schema of openser and i noticed that the >>> subscriber table only has 12 columns till rpid. >>> Looking at the openser.sql script, it has more than 12 columns >>> (including the 'version' column...). >>> What is wrong? Should SerMyAdmin add this columns to the subscriber >>> table? >>> Is it lacking me something in the database subscribet table? >>> Any help with this? >>> >>> Thanks in advance, >>> >>> Nuno >>> >>> 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> >>> >>> >>>> >>>> Hi Nuno >>>> >>>> i faced some problems your already facing and succeeded to solve them so >>>> i'd like to as you >>>> >>>> did you install SERweb tables ? >>>> >>>> if though the problems come from there especially the phplib_id field a >>>> duplication happened between it and the "id" filed so you have to remove >>>> SERweb tables and all thing will go smoothly without any errors >>>> >>>> p.s i recommend you to remove the whole database e.g openser and install >>>> it again using openser_mysql.sh script >>>> >>>> e.g /path/to/openser_mysql.sh create >>>> >>>> press yes for presence and that stuff but for SERweb no >>>> >>>> try and tell me what you get >>>> >>>> BR >>>> >>>> Muhammad >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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[Kamailio-Users] Help with SerMyAdmin
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I´m trying to install SerMyAdmin in my Openser 1.2.2 machine. I follow the steps that were on www.sermyadmin.org. I accomplished the installation task, but now when i try to do something on the DB (for example: create a new user) it gives me a lot of errors. My experience with Grails is null so i don't know were to start. Below i'll show the error messages. If anyone could give me an hand i'll appreciate. Thanks in advance, Nuno *Message:* could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Caused by:* org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Class:* UserController *At Line:* [147] *Code Snippet:* Stack org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at UserController$_closure8.doCall(UserController.groovy:147) at UserController$_closure8.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [User] at $Proxy3.saveOrUpdate(Unknown Source) ... 31 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry '4000-192.168.137.1' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 32 more

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[Kamailio-Users] Fwd: Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

---------- Forwarded message ---------- From: Asim Riaz <ariaz78(a)gmail.com> Date: Thu, Aug 21, 2008 at 2:49 PM Subject: Re: [Kamailio-Users] Help with SerMyAdmin To: Nuno Marques <yangsengopenser(a)gmail.com> my issue is same as Nano tried this command mysql -u root -p openser <openser.sql but still got ERROR 1054 (42S22) at line 1: Unknown column 'version' in 'field list' Regards Asim On Thu, Aug 21, 2008 at 12:26 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > Hi Muhammad, > > I droped the database and then created a new database as you told me... > Now i have to create all users again and an admin for sermyadmin... > But when i tried to run the mysql commands that are on > www.sermyadmin.org/openser/openser.sql they gave me an error: unknown > column 'version' in 'field list'. > Have any idea of what is going on? > By the way... do you know what happened to www.sermyadmin.org - it's seem > like it has been hacked or something... > > Many tahnks, > > Nuno > > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> > > >> >> Hi Nuno >> >> i faced some problems your already facing and succeeded to solve them so >> i'd like to as you >> >> did you install SERweb tables ? >> >> if though the problems come from there especially the phplib_id field a >> duplication happened between it and the "id" filed so you have to remove >> SERweb tables and all thing will go smoothly without any errors >> >> p.s i recommend you to remove the whole database e.g openser and install >> it again using openser_mysql.sh script >> >> e.g /path/to/openser_mysql.sh create >> >> press yes for presence and that stuff but for SERweb no >> >> try and tell me what you get >> >> BR >> >> Muhammad >> > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] Change project name in Surceforge?
by Iñaki Baz Castillo 21 Aug '08

21 Aug '08

Hi, is not time to change the project name in SourceForge? XD It appears still as "OpenSER SIP Server". -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[Kamailio-Users] SIP Session timeouts
by Jason Penton 21 Aug '08

21 Aug '08

Hi All, I have a question about the best solution to ensure that SIP calls are terminated when the 2 parties drop off the network (wireless). Is there a std way of doing this (maybe using sst module) or should this be implemented using some specialised code? If not supported in a std manner in Kamailio, i thought about creating a child process within the DLG module that continuously checks all dialogs and the connected parties and pings them for connectivity. Should connectivity of any participant drop for more than a specific unit of time then the call is focefully torn down by openser. Any ideas? Cheers Jason

2 2

[Kamailio-Users] RATELIMIT link on modules docs page?
by Alex Balashov 21 Aug '08

21 Aug '08

Why does the RATELIMIT link on http://www.kamailio.org/docs/modules/1.4.x/ go to the documentation for the Perl module? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

3 2

[Kamailio-Users] Permissions module error !
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I tried to install on my Openser 1.2.2 the permissions module so i can use an asterisk gateway. I populated the trusted table with information about the gateway. When i tried to run OpenSer it gave me some errors. Can anyone tell me what could be the problem? Below are the erros and the data on trusted table. Thanks in advance, Nuno 0(3824) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing labvoipua:/etc/openser# 0(0) permissions - initializing 0(0) Default allow file (//etc/openser/permissions.allow) not found => empty rule set 0(0) Default deny file (//etc/openser/permissions.deny) not found => empty rule set 0(0) hash_table_insert(): Unknown protocol 0(0) ERROR: permissions: trusted_reload(): Hash table problem 0(0) init_trusted(): Reload of trusted table failed 0(0) Error while initializing allow_trusted function 0(0) init_mod(): Error while initializing module permissions ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy TRUSTED TABLE +----+----------------+-------+--------------+------+---------+ | id | src_ip | proto | from_pattern | tag | version | +----+----------------+-------+--------------+------+---------+ | 1 | 192.168.137.20 | Any | ^sip:.*$ | NULL | 3 | +----+----------------+-------+--------------+------+---------+

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[Kamailio-Users] Speed dial
by salman farci 21 Aug '08

21 Aug '08

Hi, I have openser 1.2.2 installed with mysql as backend. I was able to run a simple call service after registering users in location table. Now that I'm trying implement a speed_dial, it returns "404 call not found" error at the Xlite panel after dialing a short dial number as per speed_dial table entry. But if I short dial my own short dial number it dials back to me(working) ! What might b the problem ? script file : ........... if (uri=~"sip:[0-9]{2}@.*") sd_lookup("speed_dial"); ............ speed_dial table entry : +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | id | username | domain | sd_username | sd_domain | new_uri | fname | lname | description | +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | 5 | 103 | | 33 | 10.1.2.214 | 103(a)10.1.2.214 | | | 4 | 102 | | 22 | 10.1.2.214 | 102(a)10.1.2.214 | | | 6 | 101 | | 11 | 10.1.2.214 | 101(a)10.1.2.214 | | | 9 | 105 | | 55 | 10.1.2.214 | 105(a)10.1.2.214 | | +----+------------+----------+----------------+--------------+----------------+----------+----------+-----------------+ can anyone clarify ? Thanx regards Farci

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[Kamailio-Users] Second Logo
by Darren Sessions 20 Aug '08

20 Aug '08

Good work. I tinkered a bit and came up with something very similar. - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Aug 20, 2008, at 4:52 PM, Ricky Gutierrez wrote: > logo with basic colors > > Regards > > rickygm > > ----- Original Message ---- > From: Johansson Olle E <oej(a)edvina.net> > To: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Cc: devel(a)lists.kamailio.org > Sent: Wednesday, August 20, 2008 4:55:21 AM > Subject: Re: [Kamailio-Devel] logo > > Ricky, > I really like your logo. But I also see issue with using it in pure > black/white situations - no greyshades. > > Can you try to create such a version of it as well? > > This is often needed when producing t-shirts and other stuff, where an > outline is used. > > Thanks. > /O > > <kamailio-new.png>_______________________________________________ > Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel

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[Kamailio-Users] Commercial support?
by Robert McGilvray 20 Aug '08

20 Aug '08

With the departure of voice system from the project are there any commercial entities associated with the development team that offer support for Kamailio? Regards, Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Kamailio-Users] How to test and trace the RLS messages?
by mahesh.peddi＠info-spectrum.com 20 Aug '08

20 Aug '08

Hi All, I am working on openser 1.3.2 version with RLS as "integrated_xcap_server". Its running without erros. But my query is how to test, And please send me call flow between IMS components and RLS server. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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Re: [Serusers] Content-Type: application/isup read encoded message
by Jiri Kuthan 19 Aug '08

19 Aug '08

I think the best learning experience is OOB http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… it is not really simple, on the contrary -- it is quite complete, but for that it gives you all kinds of examples that can be probably useful to you. -jiri michael(a)teldelnort.com wrote: > Hello jiri, > > Ok, > > Do yo have an idea of a simple example that will help me familiarize the > usage in ser.cfg? > > -michael > > -----Original Message----- > *From:* Jiri Kuthan [mailto:jiri@iptel.org] > *Sent:* Tuesday, August 19, 2008 03:44 PM > *To:* michael(a)teldelnort.com > *Subject:* Re: [Serusers] Content-Type: application/isup read > encoded message > > I don't think so for this particular one -- just try to use it as > other select operators. -jirimichael(a)teldelnort.com wrote:> Hello > Jiri,> > Thanks for the response.> > Is there any documentation on > how to use select operator with @context?> > > I am trying to filter > a search on the context for "II Digits:"> > -michael> > > > -----Original Message-----> *From:* Jiri Kuthan > [mailto:jiri@iptel.org]> *Sent:* Tuesday, August 19, 2008 02:09 AM> > *To:* michael(a)teldelnort.com> *Cc:* serusers(a)iptel.org> *Subject:* > Re: [Serusers] Content-Type: application/isup read> encoded message> > > I think the select operator, @content, should get you access to > the> body. There is however noparsing of SDP to my knowledge > (unless> someone corrects me, I remember it has been> > frequentlymentioned.)-jirimichael(a)teldelnort.com wrote:> Hello,> >> > How can I evaluate the Content-Type: multipart/mixed;boundary= with> > SER?> > Content-Type: application/isup; base=ansi92; version=ansi> > >> Is there a module that allows SER read the contents of the > encoded> message?> I am try to get the > > "Originating Line > Information> ----> II Digits:"> > from the message.> > -michael> > > >> > ------------------------------------------------------------------------>> > > _______________________________________________> Serusers > mailing> list> Serusers(a)lists.iptel.org>> > http://lists.iptel.org/mailman/listinfo/serusers

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[Kamailio-Users] Ratelimit outgoing SIP INVITE messages after dispatcher
by sergejf 19 Aug '08

19 Aug '08

Hello, We are connected to several VoIP carriers that impose a calls per second (CPS) rate limit on their SIP trunks. For example, on TRUNK1 we can only make call attempts at 10 CPS. Our calls are originally made by a dialer that does not support rate limiting, hence it always blasts calls as fast as it can. Upon receiving these SIP INVITE messages we determine what carrier to use and perform a mapping to a dispatcher group. Call attempts are then load-balanced between all SIP trunks from the chosen carrier. My question is whether it is possible to use the ratelimit module in conjunction with the dispatcher (or carrierroute) to ensure we don't overload the carrier SIP trunks? If this is possible, then I wonder how to configure ratelimit to support a 10 CPS call attempts rate. Thanks in advance for any advice on the topic! Serge -- View this message in context: http://www.nabble.com/Ratelimit-outgoing-SIP-INVITE-messages-after-dispatch… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] Content-Type: application/isup read encoded message
by michael＠teldelnort.com 19 Aug '08

19 Aug '08

Hello, How can I evaluate the Content-Type: multipart/mixed;boundary= with SER? Content-Type: application/isup; base=ansi92; version=ansi Is there a module that allows SER read the contents of the encoded message? I am try to get the "Originating Line Information ---- II Digits:" from the message. -michael

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[Kamailio-Users] About Asterisk2Billing
by Ali Jawad 19 Aug '08

19 Aug '08

Hi All Was anyone able to get the accounting part of asterisk2billing to work with Openser ? This looks like a pretty mature solution to me. Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] Accounting for OPenser
by Ali Jawad 19 Aug '08

19 Aug '08

Hi I am looking for a accounting solution, that I can use with Openser without involving Asterisk. Has anyone tested something similar ? I need the solution to work for my prepaid customers, for example he has got 10$ once the credit is used. Deny any more call.s With Regards Ali Jawad System Administrator

3 2

[Kamailio-Users] SIPp dispatcher load testing failure with too many hops - why?
by sergejf 18 Aug '08

18 Aug '08

Hello, I'm attempting to load test a simple dispatcher script in OpenSER 1.3.x using SIPp (built-in UAC and UAS scenarios). My dispatcher.list only has one address 8.XX.XX.12 (a SIPp instance, UAS). SIPp on 8.XX.XX.10 sends the INVITE messages to 63.XXX.XXX.110. Everything goes well until the UAC sends ACK and BYE in quick sequence. OpenSER tries to t_relay() it to itself, which results in a "too many hops" error: 2008-08-18 11:16:56:708 1219072616.708649: Aborting call on unexpected message for Call-Id '1-3548(a)8.XX.XX.10': while expecting '200' (index 8), received 'SIP/2.0 483 To Many Hops Via: SIP/2.0/UDP 8.XX.XX.10:5061;branch=z9hG4bK-3548-1-7 From: sipp <sip:sipp@8.XX.XX.10:5061>;tag=3548SIPpTag001 To: sut <sip:service@63.XXX.XXX.110:5060>;tag=2148SIPpTag011 Call-ID: 1-3548(a)8.XX.XX.10 CSeq: 2 BYE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 I have tried modifying the SIPp UAC script to re-use the Contact: header from the SIP 200 OK message in the R-URI of the SIP ACK and SIP BYE, to no avail. I have loaded the TM module but I suppose that since the SIP 200 OK is the final message of the transaction, SIP ACK and SIP BYE won't automatically be relayed to 8.XX.XX.12. Is this an OpenSER issue or a SIPp scripting issue? Any advice would be appreciated. Please find below some relevant lines from my openser.cfg: route{ xlog("TRACE:ROUTE: src($si:$sp) dst($Ri:$Rp) msg($mb)\n"); # initial checkings if ( !mf_process_maxfwd_header("10") ) { xlog("SCRIPT:ERROR: $rm (from $si:$sp) too many hops\n"); sl_send_reply("483","To Many Hops"); exit; }; if (method==CANCEL) { if (t_check_trans()) t_relay(); exit; } # routing if (has_totag()) { xlog("SCRIPT0:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); loose_route(); if (method=="INVITE") record_route(); route(1); exit; } The SIP ACK and SIP BYE follow the route defined in the above if {} statement. The SIP INVITE follows the dispatcher route shown below: record_route(); # perform load balancing # set algorithm (4 = round robin) $avp(alg) = 4; # set the group $avp(grp) = 2; if (method=="INVITE") xlog("CALL_START: RURI=[$ru] group=[$avp(grp)] callid=$ci\n"); # do balancing if(!ds_select_dst("$avp(grp)", "$avp(alg)")) { xlog("CALL_DROP:INTERNAL: no destinations for [$ru] group=[$avp(grp)], callid=$ci\n"); sl_send_reply("404", "no dst"); exit; } if(avp_check("$avp(i:273)", "eq/i:1")) { if(!avp_pushto("$ruri","$avp(i:271)")) { xlog("CALL_DROP:INTERNAL: cannot push to ruri [$avp(i:271)], group=[$avp(grp)], callid=$ci\n"); sl_send_reply("500", "cannot get dst"); exit; } } else { # redundancy t_on_failure("1"); } xlog("SCRIPT:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); # do forward route(1); exit; -- View this message in context: http://www.nabble.com/SIPp-dispatcher-load-testing-failure-with-too-many-ho… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] poll: project name
by Daniel-Constantin Mierla 18 Aug '08

18 Aug '08

Hello everybody, some people expressed inconveniences regarding the new name, so we would like to get your opinion about, therefore this is a poll that will get us to a decision. Here is the question: Do you like the name "Kamailio"? 1) Yes, I love it. Please keep it. 2) It is OK. 3) I do not mind the name at all 4) I do not like it 5) I hate the name and we should change it There is no need of explanations, as it is up to personal taste and decision. Just vote by option. Cheers, Daniel PS. for those interested... As short history about renaming story. On the 28th of March the board was informed about issues with SER part of project name. You can easily find who has the trademarks. On the 7th of April the board was informed about an agreement for the deadline as of September 30, 2008 to use old name. From there on, all old releases, documents, etc. can be referred by the old name, but nothing new. Other few discussions about what should/could be kept/changed followed and on 12th of May started the discussion about the new name with two main direction, something similar, to remind about old project or just something completely new, to be used as brand name. Couple of proposals and discussions regarding possible names followed up to 16th of July when was the deadline and the start of vote. Majority was reached on 17th but it was required to have opinions from all members of the board, last came from Bogdan, on the 28th of July. All discussions about migration were on the board mailing list. No change were done to openser.org configuration, just the new domain with kamailio in it was pointed to same server. Every step done was reported to board, following a schedule discussed there. Another discussion was about making the renaming and 1.4.0 release in same day. Being practically impossible to do all the work in one day, it was decided to do first the announcement about renaming. That happened on 28th when all agreed with the new name. As you can see, there was no 5 months of renaming duration. There was no change in openser.org DNS configuration undertaken with/without the knowledge of all board members. The new name domains were reserved when majority was achieved, in any case; dns entries for new name were pointed to old server shortly after booking the domains. -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] SIP and IP multicast
by Steven C. Blair 18 Aug '08

18 Aug '08

Has anyone had any experience using SIP via SER to setup an IP multicast session with a group of phones?

2 1

[Kamailio-Users] Looking for a VOIP Engineer
by Jonathan K. Creasy 18 Aug '08

18 Aug '08

Voxitas Inc. is a national VOIP provider based in St. Louis, MO. and we are currently looking for qualified applicants to fill Tier Three support positions and a VOIP Engineer position. The reason I am posting to this list is because we are looking to fill the VOIP Engineer position with someone who has OpenSER experience and is comfortable configuring and maintaining these systems. For details please contact our VP of Operations Zack Sargent. Zack Sargent VP Operations and Engineering Voxitas, Inc. 210 N. Tucker, St. Louis, MO. 63101 314-266-4000 x108 zsargent(a)voxitas.com Thank you! -Jonathan

1 0

[Kamailio-Users] PortaBilling
by Ali Jawad 18 Aug '08

18 Aug '08

Hi All Has anyone tried portabilling with Openser ? If so can you please provide a sample of your config file ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

1 0

[Serusers] SER - Waiting Transaction
by Arun Kumar 18 Aug '08

18 Aug '08

Hi, I'm running SER 0.96 in my production for a long but there is strange problem , when I do serctl moni and looks for transaction it shows waiting transaction in () almost 400+ can some one tell me why there are so many transaction in waiting while I see processing only 10-20 transaction. Thanks arun

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[Kamailio-Users] pick_branch failed (lowest==-1) for code 302--HELP!!
by Krunal Patel 18 Aug '08

18 Aug '08

Hi, Here is the scenario: -----------------------Invite--------------> <----------100 Trying---------------------- <----------302 Moved temporarily-------- -----------------------ACK--------------> Now I would like to route call to contact returned in 302 responce. But I am getting problem: Aug 17 18:10:41 [16667] DBG:core:parse_headers: flags=80 Aug 17 18:10:41 [16667] DBG:tm:t_should_relay_response: T_code=100, new_code=302 Aug 17 18:10:41 [16667] DBG:tm:t_pick_branch: picked branch -1, code 999 Aug 17 18:10:41 [16667] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 302 Aug 17 18:10:41 [16667] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Please help me to fix this. Thanking you Krunal Patel

1 1

[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 17 Aug '08

17 Aug '08

I will be out of the office starting 08/15/2008 and will not return until 08/18/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

1 0

[Serusers] Acc module extra patches
by caio 16 Aug '08

16 Aug '08

Hello, Is there patches for acc extra params for ser version 0.9.7? I've found this link http://www.voice-system.ro/docs/acc-headers/ but do not know if is it for v0.9.x Thanks... -- caio

1 0

[Serusers] on Ubuntu Installation & Config
by Ameed Jamous 16 Aug '08

16 Aug '08

Hi all, Am trying to install & Configure SER on ubuntu 8.04. After i intsalled the SER file i can see this in my terminal: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.104 [192.168.1.104]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.104 [192.168.1.104]:5060 Aliases: tcp: Ubuntu-AM.local:5060 tcp: localhost:5060 udp: Ubuntu-AM.local:5060 udp: localhost:5060 ok now how can i add a user to register to SER ? How can i monitor the logs and the prosses happening? i tried using serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config Even though i have line fifo=/tmp/ser_fifo in my config. Can anyone help with this? Regards, Ameed -- Best Regards, Ameed Jamous Project Manager/Carriers Relations. Direct Line: +962 777 835819. www.joodtelecomgroup.com

1 0

[Serusers] call forward no answer
by caio 15 Aug '08

15 Aug '08

Hi all, I can't make call forwarding on no answer works on my ser-0.9.7 installation. Here you can see a snippet of log, where the call falls on failure_route and then to route_1 where t_relay is called. Log at: http://rafb.net/p/xwTmn195.html The call is from sip:579951 to sip:579915(who has enabled the feature fwdnoanswer to sip:579916. But I do not know why the INVITE after the append_branch() is still to R-URI: sip:579915@.... After the relay then the call terminates with: Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ I follow the example of ser guide.., but do not work.. I would appreciate any comment.. Regards, -- caiogf

1 0

[Kamailio-Users] Question
by Edgar Guadamuz 15 Aug '08

15 Aug '08

Hello, I'm a newbie and I don't know the difference between opensips (where openser.org redirects) and kamailio, If someone can give me a light on this it would be great, thanks,

3 2

[Kamailio-Users] project foundation
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello everybody, I think it is the time to move forward with project management structure. During last days there were talks and discussions on the lists, the solution to make the project independent and long term trustable is to create a foundation. We made a draft of purpose and organization structure and we want to get your comments. http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft For the statute we are looking at similar cases of foundations around open source projects. Considering the developer population and contributors, I find as good and reliable place to register the foundation to be Germany. There is still long way to go and a lot to discuss: * membership fees * how one can become a full member (associate member can be anyone that is friend of the project and pays the membership fee) * how a member loses the board or full member rights * how the board is supervised o when and how the reports of the board are presented o when and how the general assembly meeting takes place o how the reports are approved by full members * how to change the statute over time ... and perhaps more, but if we start and put some effort we can have it ready in reasonable time, avoiding trust and reliability issues in long term. Let's have debates here and contribute the results to the wiki. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] project status and my decisions
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello all, following the last events everybody perhaps wonders what is going on. I was surprised by the way everything happens and I am not sure yet how this resulted. Probably a bit of time is needed to clarify relations between developers themselves and community as well. There is not any official statement that a developer quited the support for this project, but I guess the trust between some is at lowest level possible. It is clear that we cannot say anything about a board right now that manages the project. From board discussions summary you may see that discussions were conducted to create a legal entity around the project, a non profit foundation. This might be hard to achieve in this status, but as a personal decision, I am not going in any other management group unless there is a legal framework backing up. Human relations prove to be pretty unpredictable and not conforming with initial statements. What is sure now - openser 1.3.x and 1.4.x will be maintained properly and I am volunteering to take care but hope others will join. It is really a good achievement of a sip server. Decision about future development will be based on community and developers feedback for this project. I am not going to promise anything by myself as I admit one person (to be more specific: myself) cannot take care of the entire code base -- I am just realistic, not willing to sell dreams. I think the people that still believe in the project can come with suggestions and alternatives of future development. What I can tell for sure, I am not going to join a fork of the original openser project. If something happened to make developers not working together, solution to solve nicely could have been found, in the worst, maybe just by splitting teams, but keeping old domain together just to show the new alternatives. "Why not joining a fork in this manner?"> Because I was one of the founders of this one and involved in development and management at top level. If I failed to build something I started, I had all the tools in my hands, why just creating something similar saying the old is crap but this new one will be the best? What is the difference? Why I didn't do it first time? Would I be able to fool someone? Does leaving my ship in secrecy on back door will bring more trust on me? I doubt and perhaps you, too. I am certainly assuming share of the failure in managing this project. I was there, in management, period. There is no single person responsible for mis-management, but it comes just easy for some to blame the others. But I am really confident we built outstanding sip server, all of us contributed with code, documentation, testing to an application that routes million and millions of minutes per day. There were accuses of bad things about project, I will argument in a separate message, this will get too long. So, I am asking: - how do you see a proper management structure for the future - feedback about future involvement, suggestions and alternative for going on Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] Register to SER with UDP
by Steven Wu 15 Aug '08

15 Aug '08

Hi All, Does SIP server at iptel.org support UDP register? When my UA rigister to iptel.org whith UDP, the 401 response always replied by SER at port 100xx(example 10028). But in my contact of rigister port assigned to 5060. So, my UA can't recive the 401 response, register failure. But use TCP everyting is OK? Can anyone give me help? Best regards, Steven Wu

2 1

[Kamailio-Users] KAMAILIO v1.4.0 is released
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

KAMAILIO v1.4.0 is released August 7, 2008 Kamailio is the new name of the SIP server known so far as OpenSER. Due to trademark issues, the OpenSER project had to change the name. Officially, the new name was announced on July 28, 2008. Version 1.4.0 brings many new features since OpenSER v1.3.0 as well as improvements to existing components. An important share of effort was directed to code cleanup and stability. Moreover, new features in core and old modules were added while other 6 new modules were introduced. Download: You can download the tarball of the released sources at http://www.kamailio.net/pub/kamailio/latest/src/ Binary packages for several distributions you can find at http://www.kamailio.net/pub/kamailio/latest/bin/ http://www.kamailio.net/pub/kamailio/latest/packages/ Packages will be uploaded as soon as they are submitted. Documentation: * modules' documentation: http://www.kamailio.net/docs/modules/1.4.x/ * migration guide : http://www.kamailio.net/dokuwiki/doku.php/install:1.3.x-to-1.4.0 Summary of the new stuff in core: * overhauled DB API - better performance and safety, more common functionality integrated in the core * extensive cleanups in database drivers - integrate common functionality into the core for more stability and maintainability * SDP parser - provides an internal API for parsing SDP * fixup functions - provides an internal API for fixing module parameters * new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl * extension to module interface - its now possible to use up to 6 parameters in module functions New modules: * db_oracle - ORACLE DB driver - connect to Oracle SQL Server * dialplan - Dialplan management - dialplan regular expression based translations * nat_traversal - NAT traversal helper * peering - RADIUS based peering allowing SIP providers to verify via a broker if source or destination request is from a trusted peer * ratelimit - SIP ttraffic shaping control and server load control based on your rules * userblacklist - User-specific blacklists New in existing modules: * acc module * it is possible to log accounting related output to a different log facility * cfgutils module * new FIFO functions get_config_hash and check_config_hash for config file checks * shared PV exported to configuration script * debugging helpers usable from config script for abort, shm_status and pkg_status * mysql module * enable timeouts on connect, read and write to prevent blocking on errors * database modules * renaming of all database modules, addition of the prefix 'db_', e.g. mysql is now db_mysql * carrierroute module * extension: much more flexibility in routing and database supported failure routing, improved internal structure * dialog module * internal API reworked for better flexibility * direction of the message provided via the internal API * new dialog callback types * new mi command: dlg_list_ctx * dispatcher module * ability to load destination URI flags from database at startup or reload * new algorithm to hash the content of a pseudo-variable * enum module * enum_fquery([...]) replaced by enum_pv_query([...]) * msilo module * notification system refurbished - message body, content-type, from address and contact header can be dynamically specified with pseudo-variables * pdt module * accepts now same destination domain bound to different prefixes * internal structures and logic optimized for memory and performance * lcr module * refactor module functions * pseudo-variable support * mediaproxy module * update to version 2.0 * better performance and scalability as packets are forward in kernel space New testing suite * a testing suite to ensure quality checking and regression tests reports has been developed, included in the repository and used during this development cycle Other important changes * modules' documentation has been migrated to DOCBOOK XML format * doxygen documentation extended and developer guide published * serweb tables have been removed * command line tool to manage dbtext in a SQL fashion These are not all, there is a big change log that gives more details: * http://www.kamailio.net/pub/kamailio/latest/ChangeLog More on http://www.kamailio.net Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] As a board member
by Adrian Georgescu 15 Aug '08

15 Aug '08

Hello, Because of sheer incompetence and abuse conduct of one of the recently adopted board members I decided to put my activities as a management board member of OpenSER/Kamailio project on hold. I will be able to resume my activities after the voluntary leave of Henning Westerholt. For those who do not know who I am, I was involved with OpenSER from Day 0 of its inception. Kind regards, Adrian

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[Serusers] SIP conference.
by Peter Cai 15 Aug '08

15 Aug '08

Hi all! I know H323 support conference but SIP says nothing about it. I want to make some study on this subject but there seems to be little information on Internet. Could some one give me some clues? Is there any RFC draft on this concept?

2 1

[Kamailio-Users] Voicemail implementation...
by Gerard A. Matthew 14 Aug '08

14 Aug '08

Ok, I'm trying to get the voicemail system working, however for some reason my Asterisk box is receiving a duplicated IP Address on the c line from OpenSER....below is the sample sip header. Is there anything in my openSER config file that would cause such a duplication? Rgds, Gerard INVITE sip:u5073@<DOMAIN>:5061 SIP/2.0 Record-Route: <sip:<DOMAIN_IP>;ftag=1d498377;lr=on> Via: SIP/2.0/UDP <DOMAIN_IP>;branch=z9hG4bKaed4.96adae34.1 Via: SIP/2.0/UDP 192.168.1.112:51006;received=<CLIENT_XIP>;branch=z9hG4bK-d8754z-0127430e81514e54-1---d8754z-;rport=51006 Max-Forwards: 69 Contact: <sip:0078@<CLIENT_XIP>:51006;nat=yes;nat=yes> To: "5073"<sip:5073@<DOMAIN>> From: "G Laptop"<sip:0078@<DOMAIN>>;tag=1d498377 Call-ID: NDZiOTUyNWFmYTQ3M2JlMjNiYzdmZjcyMGYyZDljZWE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 481 P-hint: usrloc applied v=0 o=- 6 2 IN IP4 192.168.1.112 s=CounterPath X-Lite 3.0 *c=IN IP4 <XXX.XX.XXX.XXX><XXX.XX.XXX.XXX>* t=0 0 m=audio 3649036490 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 1 : 4IoIRnGz iYQAZz1E 192.168.1.112 16720 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes

2 1

[Kamailio-Users] Translate name in RURI to number
by Ali Jawad 14 Aug '08

14 Aug '08

Hi all..my users are all registered with names...I want to forward their calls to my pstn gw..however those only take numerical ANI numbers. So I need to change the sip packet on the Openser server to a number (diff number per user) before forwarding that to the PSTN Gateway ..any ideas ? Would ENUM work ? THanks

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[Kamailio-Users] [OT] Any domain registrar offering SRV, NAPTR and AAAA?
by Iñaki Baz Castillo 14 Aug '08

14 Aug '08

Hi, first of all I'm sorry for the OffTopic. I'm looking for a domain registrar and nameserver that offers me SRV, NAPT and AAAA registers ni their domain control panel. Unfortunatelly in Spain *NO ONE* offers it. Is there any one "outside"? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[Kamailio-Users] Kamailio & CDRTool 6.5.XX & MediaProxy 2.0
by luzango mfupe 14 Aug '08

14 Aug '08

Hi guys,I read the release note of MediaProxy 2.0 that to be able to have it fully functioning one needs CDRTool version 6.5.X and above i have been searching all over with no avail can someone point where it if found?? by the way i just found the old version 6.4.1 in the author's site wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

1 0

[Kamailio-Users] new developer: Carsten Bock
by Daniel-Constantin Mierla 14 Aug '08

14 Aug '08

Hello, I wish to welcome long time contributor, Carsten Bock, among developers. He is going to be officially co-maintainer of the dispatcher module and help with several other modules such as dialog, uac, ... He has submitted lot of new features in the past to many parts of the code in openser. His id is 'carstenbock'. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] openser 1.2.1 errors in the log
by toly 14 Aug '08

14 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS support. TLS is not turned on Did anybody experienced with the errors listed below: 08/13 16:28:05 13648 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one 08/13 16:28:05 13651 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 08/13 16:28:05 13651 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/13 16:28:05 13651 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/13 16:28:05 13651 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/13 16:28:05 13651 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/13 16:28:05 13651 debug <t_funcs.c: 255> ERROR:tm:t_relay_to: t_forward_nonack returned error Once I start getting these, proxy will become pretty much un-usable, I have to restart it. Thanks a lot. Hippo -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp18973275p18973275.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Error: driver error on query: Unknown column 'str_doc_col' in 'field list' ??
by David Loh 14 Aug '08

14 Aug '08

Hi guys, Since 1.4.0 released on Kamailio, I've been testing it on my R&D server. With the introduction of RLS, I'm trying to implement this module along with presence on my box, however, I'm having this error messages whenever rls_handle_subscribe() been called. ------------------------------------------------------------------------ Aug 14 00:06:03 localhost /sbin/kamailio[5139]: INFO: [sip:davidloh@192.168.72.128](192.168.72.1) - [eyeBeam release 1100l stamp 46319] entered Aug 14 00:06:03 localhost /sbin/kamailio[5139]: DBG: Request [SUBSCRIBE] from [sip:davidloh@192.168.72.128] (192.168.72.1) Aug 14 00:06:03 localhost /sbin/kamailio[5139]: _ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'str_doc_col' in 'field list'_ Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:core:db_do_query: error while submitting query Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:get_resource_list: while querying table xcap for [uri]=sip:davidloh@192.168.72.128 Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:rls_handle_subscribe: while attepmting to get a resource list Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:rls_handle_subscribe: occured in rls_handle_subscribe ------------------------------------------------------------------------ Have been google around for the solution but couldn't find any. Appreciate if anyone could enlighten me on this. Meanwhile, I was wondering if any of could enlighten me on how to get Kamailio + Presence + RLS works? No matter how I configure, SIP client (eyeBeam) seems not able to pull out any buddies-list/anything right from XCAP (integrated XCAP from Kamailio). Following is my configuration (unnecessary part have been striped) ------------------------------------------------------------------------ loadmodule "sl.so" loadmodule "tm.so" loadmodule "db_mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 0) modparam("presence_xml", "integrated_xcap_server", 1) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "server_address", "sip:192.168.72.128:5060") loadmodule "imc.so" loadmodule "mi_xmlrpc.so" modparam("mi_xmlrpc", "log_file", "/var/log/kamailio-xmlrpc.log") modparam("mi_xmlrpc", "port", 8080) loadmodule "pua.so" loadmodule "rls.so" modparam("rls", "integrated_xcap_server", 1) modparam("rls", "xcap_root", "http://192.168.72.128/xcap-root:8080") modparam("rls", "rls_event", "presence") modparam("rls", "to_presence_code", 10) loadmodule "xcap_client.so" modparam("xcap_client", "xcap_table", "xcap") modparam("xcap_client", "query_period", 50) modparam("presence|presence_xml|imc|pua\rls", "db_url", "mysql://myuserid:mypassword@localhost/kamailio") ........... route[0] { .... if (is_method("NOTIFY")) { pua_update_contact(); route(4); } else if (is_method("PUBLISH|SUBSCRIBE")) { route(5); } route[5] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { $var(ret_code) = rls_handle_subscribe(); if($var(ret_code)==10) { handle_subscribe(); } t_release(); } else if(is_method("NOTIFY")) { rls_handle_notify(); xlog("L_INFO", "DBG: RLS handle_notify()"); } exit; } ------------------------------------------------------------------------ Did I missed anything ? Thanks, David Loh

2 1

[Kamailio-Users] Non-numeric prefixes in carrierroute:scan_prefix
by mayamatakeshi 14 Aug '08

14 Aug '08

Hello, I just need a confirmation. We have Kamailio 1.3 running in production and it is working fine. But we received some reports from our OAM team that sometimes they want to set a non-numeric prefix at carrierroute::scan_prefix (like 'abc.') and this seems to not work. I have no complete details but it seems cr_user_rewrite_uri will fail if scan_prefix holds a non-numeric prefix. So I just like to know if scan_prefix is supposed to work with non-numeric prefixes or not. Regards, takeshi

2 1

[Serusers] SER with Radius Authentication
by Samuel Muller 13 Aug '08

13 Aug '08

Hello all, I'm looking for an updated sample of "ser.cfg" who can help me about the use of Radius for authentication. Since release 2.0.0, it seems the function "www_challenge" does not exist ... If anyone has a config part concerning Register + Auth with Radius, it could be very helpfull ! In second point, I just wanna know if a database is always needed for multi-domains environment ? I've some problems with functions like lookup_domain. Many thanks, Samuel MULLER

1 0

[OpenSER-Users] Problem getting redirect reason in uac_redirect
by Ruchir 13 Aug '08

13 Aug '08

I'm using uac_redirect module to handle redirect and accounting from openser. Redirect works fine so as cdr but I'm not getting how to write redirect reason in cdr. I didn't find any way to find and store redirect reason(call forward, busy, no answer) in CDR. Does anyone know how to do it?

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[Kamailio-Users] ERROR:uri_db:mod_init: Invalid table version of the subscriber table (Kamailio ver. 1.4)
by Rafael Risco Gonzales-Vigil 13 Aug '08

13 Aug '08

Hi I am trying to run new kamailio 1.4 but have this errors: *Aug 11 12:56:44 [20808] ERROR:uri_db:mod_init: Invalid table version of the subscriber table Aug 11 12:56:44 [20808] ERROR:core:init_mod: failed to initialize module uri_db Aug 11 12:56:44 [20808] ERROR:core:main: error while initializing modules* I have created new database 'kamailio' using kamdbctl and 'version' table has this values: table_name: subscriber table_version: 6 any idea? rafael PS: same problem with opensips 1.4...

2 4

[Kamailio-Users] listen the new name
by Daniel-Constantin Mierla 13 Aug '08

13 Aug '08

Hello, we did some recordings of the new name, letting you know how some people pronounce the word. The audio files are linked from the first page http://www.kamailio.net/ or can be found directly at: http://www.kamailio.net/pub/kamailio_pronounciation/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] project foundation
by Daniel-Constantin Mierla 12 Aug '08

12 Aug '08

Hello, On 08/12/08 16:39, Adam Linford wrote: > Good stuff! > > I think that, for the most part, the business and technical facets of > the organisation should remain fairly distinct. The board needs > representation with regards to technical direction (and should have > this input from the organisations official 'members' under my > understanding of the current proposal), but its purpose should be to > keep developers free to code, and users free to use, safe in the > knowledge that resources will be made available (where neccessary) for > the project to achieve its objectives (both long and short term). yes, there are going to be no restrictions in usage or coding, but have an organization with legal framework behind that can take care of trademarks, manage resources and administration of the project. > > We could make non-executive positions on the board available. A good suggestion. > These should be unbiased people.. perhaps invite some big names in the > SIP field, that can provide an independent review of the boards > activities, as well as assisting the board in its role. This is a > tried and tested directorial structure that would also appeal to the > corporate world, which in turn might make it easier to achieve > corporate sponsorship/contributions/membership (as well as adoption of > the software in industry!). > > I don't think an executive committee is needed within the board.. the > board itself should suffice. If it doesn't, we don't have the right > board. I would suggest keeping the main board size small (5-8) to > keep it agile whilst maintaining good representation of members I think would be good to have some clear positions so people should know who to address to when comes to dealing with the foundation. Also, by this, people can allocate the proper time in their schedule for the tasks. We may avoid in this way some members expecting the others to do some work. > > Election rules need discussing.. duration of term for board and also > the achievement of 'member' status.. we dont want potential developers > discouraged from working with us in the traditional open source model > because they would have too much responsibility placed on them to be > 'active members'. Being an active member is just free will. But we want to avoid being a member with voting right just by paying some money. That meber should show interest and contribute to the project in a positive manner. Indeed, this needs to be addressed. > > Frequency of meetings etc are all relatively easy decisions that can > be deferred to the board once i elected. Yes, I believe the same. Cheers, Daniel > > Cheers, > Adam > > On 12 Aug 2008, at 12:16, Daniel-Constantin Mierla wrote: > >> Hello everybody, >> >> I think it is the time to move forward with project management >> structure. During last days there were talks and discussions on the >> lists, the solution to make the project independent and long term >> trustable is to create a foundation. >> >> We made a draft of purpose and organization structure and we want to get >> your comments. >> >> http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft >> >> For the statute we are looking at similar cases of foundations around >> open source projects. Considering the developer population and >> contributors, I find as good and reliable place to register the >> foundation to be Germany. >> >> There is still long way to go and a lot to discuss: >> >> * >> >> membership fees >> >> * >> >> how one can become a full member (associate member can be anyone >> that is friend of the project and pays the membership fee) >> >> * >> >> how a member loses the board or full member rights >> >> * >> >> how the board is supervised >> >> o >> >> when and how the reports of the board are presented >> >> o >> >> when and how the general assembly meeting takes place >> >> o >> >> how the reports are approved by full members >> >> * >> >> how to change the statute over time >> >> ... and perhaps more, but if we start and put some effort we can have it >> ready in reasonable time, avoiding trust and reliability issues in long >> term. >> >> Let's have debates here and contribute the results to the wiki. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] branch_route not always called
by Alex Hermann 12 Aug '08

12 Aug '08

Hello again, The uac_replace_from issue led me to another problem. The branch_route is not always called even though I set it using t_on_branch everytime from REQUEST_ROUTE and FAILURE_ROUTE. The failing scenario is as follows: The call is routed to 2 domains (A,B) each with 2 SRV records (1,2). t_on_branch relay domainA branch domainA-1 - timeout branch domainA-2 - 480 failure route append_branch t_on_branch relay domainB branch domainB-1 - timeout branch domainB-2 - 200 OK ==> this branch doesn't call the branch_route Any idea's ? -- Greetings, Alex Hermann

2 3

[Serusers] problem error SER
by Bimantara Jayadiputro 12 Aug '08

12 Aug '08

hello sir i'm bimbim, student from indonesia now i'm trying to finish my final project my final project is about SER i have trouble with that when i start /etc/init.d/ser start there is an error with assigning addres below is the configuration of my ser ------------------------------------------------------------------------------------------------------------------------ cnc9@cnc9-desktop:/var/run/ser$ sudo /etc/init.d/ser start Starting ser: serListening on udp: 10.14.200.95 [10.14.200.95]:5060 udp: 2001:660:4701:1:0:0:0:1 [2001:660:4701:1:0:0:0:1]:5060 udp: sip4.sipv6.net [10.14.200.101]:5060 udp: sip6.sipv6.net [2001:660:4701:1:0:0:0:2]:5060 tcp: 10.14.200.95 [10.14.200.95]:5060 tcp: 2001:660:4701:1:0:0:0:1 [2001:660:4701:1:0:0:0:1]:5060 tcp: sip4.sipv6.net [10.14.200.101]:5060 tcp: sip6.sipv6.net [2001:660:4701:1:0:0:0:2]:5060 Aliases: tcp: localhost:5060 tcp: cnc9-desktop.local:5060 udp: localhost:5060 udp: cnc9-desktop.local:5060 stateless - initializing 0(0) . cnc9@cnc9-desktop:/var/run/ser$ Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 * 0(0) ERROR: udp_init: bind(6, 0x80f8d7c, 16) on 10.14.200.101: Cannot assign requested address* ------------------------------------------------------------------------------------------------------------------------ i also attached my ser.cfg i hope you can help me resolve this problem thank you

3 2

[Kamailio-Users] uac_replace_from and lumps again
by Alex Hermann 12 Aug '08

12 Aug '08

Hello, i'm still fighting with the very annoying lump model that OpenSER uses. I need to send out INVITE's with different From headers in a failover situation. More specifically I need to send out each step -original request and failure_routes- with a different From header. If any request forks (parallel or serial due to DNS SRV or registered contacts), it needs to keep current (same) the replacement From on all branches. I tried the following: try 1: REQUEST_ROUTE: uac_replace_from(a) t_relay(); FAILURE_ROUTE: append_branch() uac_replace_from(b) t_relay() the second uac_replace_from always produces garbage in the From try 2: REQUEST_ROUTE: t_newtran() uac_replace_from(a) t_relay(); FAILURE_ROUTE: append_branch() uac_replace_from(b) t_relay() Now the second uac_replace_from is correct, but when the first t_relay() does dns based failover, the original from is used for the second branch instead of the replacement. If the response code is locally generated 408, the second uac_replace_from produces garabage too. Is someone aware of a working solution for this (very common) situation? -- Greetings, Alex Hermann

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[Kamailio-Users] Need Help Configuring OpenSer with OpenXCAP
by Noble Antony T 12 Aug '08

12 Aug '08

Hi, I could configure Openser as RLS, and installed OpenXCAP. I wanted to use OpenXCAP with EyeBeam Client. (version 1.5.10.2 build 33793) OpenXCAP is giving the following error. Invalid Document Selector xcap-root(a)192.168.176.150/pres-rules/users/user1/presrules (the Document Selector context must be 'users' or 'global': 'pres-rules') Invalid Document Selector xcap-root(a)192.168.176.150/resource-lists/users/user1/resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') Invalid Document Selector xcap-root(a)192.168.176.150/pres-rules/users/user1/presrules (the Document Selector context must be 'users' or 'global': 'pres-rules') Invalid Document Selector xcap-root(a)192.168.176.150/resource-lists/users/user1/resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') UE is getting 404 Error. It seems the OpenXCAP is expecting 'users' or 'global' instead of 'resource-lists'. Can somebody please point out, if I am missing something..? I am attaching my Config Files along with this. Somebody please help me... -- Thanks & Regards, Noble Antony Thattil.

1 1

[Kamailio-Users] fr_inv_timer increased to 120 secs
by Kaeman Chris 12 Aug '08

12 Aug '08

Hello, What are the problems in production that can be encountered if the fr_inv_timer is increased from 30-45 secs to 120 secs for a sip-sip call? Anyone have found any issues? Thanks! KC --------------------------------- Unlimited freedom, unlimited storage. Get it now

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Re: [Kamailio-Users] ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2
by Daniel-Constantin Mierla 12 Aug '08

12 Aug '08

Hello, On 08/12/08 10:30, mahesh.peddi(a)info-spectrum.com wrote: > Hi Daniel, > > Thanks for your prompt reply.The output of "show create > table version" is as follows > > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > | Table | Create Table | > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > | version | CREATE TABLE `version` ( > `table_name` varchar(32) NOT NULL, > `table_version` int(10) unsigned NOT NULL default '0' > ) ENGINE=MyISAM DEFAULT CHARSET=latin1 | > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > 1 row in set (0.03 sec) > > Kindly guide me on this. the table structure is ok. Could you make sure the mysql module is compiled again if you updated libraries or so. I was pointed that there was a change in naming the INT type in mysql libraries -- maybe others on the list can jump quicker here with a yes or no. Cheers, Daniel > > Thank you, > > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: <users(a)lists.kamailio.org> > Sent: Monday, August 11, 2008 9:41 PM > Subject: Re: [Kamailio-Users] ERROR:presence:mod_init: 1Wrong version > v-1 for table <presentity>, need v2 > > >> Hello, >> >> please post the output of: >> >> show create table version >> >> in the mysql client. The errors shows some issue with the column type. >> >> Cheers, >> Daniel >> >> On 08/11/08 17:12, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> >>> I am working on Openser 1.3.2 version with no_tls support. >>> When i include >>> "modparam("presence", >>> "db_url","mysql://openser:openserrw@localhost/openser")" >>> >>> for the presence and run openserctl moni i am getting the error as >>> >>> ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo >>> ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", >>> "/tmp/openser_fifo")' in your config >>> ERROR: and also have loaded the mi_fifo module. >>> >>> And the errors in /var/log/messages >>> >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:table_version: >>> invalid type (2) or nul (0) version columns for presentity >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:presence:mod_init: 1Wrong >>> version v-1 for table <presentity>, need v2 >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:init_mod: failed to >>> initialize module presence >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:main: error while >>> initializing modules >>> Aug 10 09:37:41 rhel4 openser[5530]: NOTICE:presence:destroy: >>> destroy module ... >>> Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session opened for user >>> root by (uid=0) >>> Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session closed for user >>> root >>> >>> And when i check in the data base its showing the version as >>> >>> mysql> select * from version where table_name="presentity" ; >>> +------------+---------------+ >>> | table_name | table_version | >>> +------------+---------------+ >>> | presentity | 2 | >>> +------------+---------------+ >>> 1 row in set (0.00 sec) >>> >>> Please help me how to resolve this. >>> And as mention in the forums i modify the res.c as below shows even >>> i am facing the same error. >>> >>> >>> --------------------------------------------------------------------------------Index: >>> modules/mysql/res.c===================================================================--- >>> >>> modules/mysql/res.c (revision 4448)+++ modules/mysql/res.c (working >>> copy)@@ -103,7 +103,10 @@ break; default:- RES_TYPES(_r)[i] = >>> DB_STRING;+ LM_WARN("unhandled data type column (%.*s) type id (%d), >>> "+ "use DB_STRING as default\n", RES_NAMES(_r)[col]->len,+ >>> RES_NAMES(_r)[col]->s, fields[col].type);+ RES_TYPES(_r)[col] = >>> DB_STRING; break; } >>> }--------------------------------------------------------------------------------I >>> am enclosing my "openser.cfg" and "openserctlrc"Regards,Mahesh >>> PeddiInfospectrum India Pvt. Ltd.Cell: +91 9765775176IP-Phone Ext. >>> - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] poll: project name
by Helmut Kuper 12 Aug '08

12 Aug '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Klaus, Of course I agree. It's a matter of time to know what Kamailio stand for. I think Apache is a nicer name than Kamailio, but anyway: I agree the name is mainly a question of personal taste and currently I don't like that taste - nevertheless, I think the name should be changed to a more handy one, just to round up the whole product. And sure: The name won't stop telcos from using the Software rather that project structure, roadmap and quality. regards and peace Helmut Klaus Darilion schrieb: | Helmut Kuper wrote: |> -----BEGIN PGP SIGNED MESSAGE----- |> Hash: SHA1 |> |> Hm, |> |> well, maybe I have to make it a bit more clear what I meant. Kamailio is |> pronounceable in german, but it is not easy in terms of smoothyness - |> like openser was or freeswitch is ... |> |> KAMAILIO hmmm no, not with the best will in the world: I have problems |> with it to say it without explaining what that name points to. I think |> it is not a good name to inspire confidence. Maybe it needs a slogan to |> explain a bit more :D | | IMO this is all a matter of time. Would you think that Apache is a web | server and Eclipse is an IDE and Asterisk is an IP PBX? Me not. But now, | if you tell one of these names to an IT guy they know what you are | talking about. Thus, in my opinion, we just need some trust in the | product - I do not believe the argument that telcos will use open source | only if the name indicates open/free/SIP.... Nobody uses a product just | because of its name. | | But I have to accept that others don't like name ...:-( | | regards | klaus | | -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkihQDkACgkQ4tZeNddg3dwrJgCdENGNK/PAid+ftRtvgUFyP9hY ZXAAoJ5ErmBLbKuxTRvp6WmWzudI1BS9 =cyPy -----END PGP SIGNATURE-----

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[Kamailio-Users] Identifying branch in ONREPLY-ROUTE?
by Alex Balashov 12 Aug '08

12 Aug '08

What is the simplest way to identify the branch (by branch index) to which a reply belongs in an ONREPLY-ROUTE? I have parallel forking with my registrar and do passive call statekeeping. The problem is that it must be branch-aware; I need to know if, for example, I got a successful provisional reply on one branch of the call and a failure on the other, and deal with those separately. Right now my statekeeping information is keyed by Call-ID, but since that persists across two branches, obviously some sort of branch-identifying information has to be present. I've tried setting a persistent variable to the value of $T_branch_idx inside a BRANCH-ROUTE, but in ONREPLY-ROUTE that just shows up with the most recent branch index, regardless of which branch the reply actually came in on. I'm probably missing something quite obvious here, so I ask - how do I incorporate some form of branch awareness into ONREPLY-ROUTEs? I really do not care what form that is - be it branch index, or some other built-in value, as long as it is consistent. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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Re: [Kamailio-Users] simple question: log 180, 200 responses
by Alex Balashov 12 Aug '08

12 Aug '08

Ali Soltani wrote: > Hi Alex, > > Thank you for the response. > > I tried to use onreply_route block in the main route as well as in > route1 as follow but still can not log response messages (180, 200, ..) > > route > { > t_on_reply("1"); > ... > } > > route[1] > { > if (is_method("INVITE")) > { > t_on_reply("1"); > } > ... > } > > onreply_route[1] > { > xlog("(onreplyroute1) usr=$ru method=$rm request_port=$rp > received_port=$Rp original port=$op destination_port=$dp src_port=$sp"); > } You can only have one reply route per request. If you specify multiple ones, only the last one will take effect. The problem in this case is that route[1] will never be called. Here's what you want: route { if(is_method("INVITE")) t_on_reply("1"); } onreply_route[1] { ... } -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] free_fixup_function
by Julien mangeard 12 Aug '08

12 Aug '08

In the 1.4 release notes, I have read that: => new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl I don't find any reference to "free_fixup_function" in the Perl module source code. Are you sure that this feature works for Perl module? Someone work on it? Best Regards, -- Julien Mangeard

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[Serusers] Fw: 20080812 - Need help with configuring SER as a call forwarder server
by Morteza Alizadeh 12 Aug '08

12 Aug '08

Dear Sir/Madam, My previous email has been held by your system due to its large volume. I removed the log file from it and resend it again. If you think, that logs of the SER would help you, please provide me the way to send it as well. Your help is much appreciated. --- Sincerely yours, Morteza Alizadeh Design Engineer Cell Phone: +60 (12)-319-7439 Email: malizadeh(a)emerj.com Morteza.Alizadeh(a)gmail.com Skype: mortezaalizadeh Block D-17-6, Gurney Height Condominium, Jalan Bukit Keramat, 54000 Kuala Lumpur, Malaysia ----- Forwarded Message ---- From: Morteza Alizadeh <malizadeh(a)e-merj.com> To: serusers(a)lists.iptel.org Cc: vazirif(a)e-merj.com Sent: Tuesday, August 12, 2008 10:09:35 AM Subject: 20080812 - Need help with configuring SER as a call forwarder server Dear Sir/Madam, I am trying to use and configure your SER in a manner to let us forward our landline calls to the public sip.broadvoice.com gateway. Unfortunately, I am completely mixed up with the way it should be setup. There are some questions that I would be really appreciated if you can answer them or forward it to a right person. Please consider that I have already fully configured your SER to handle internal IP call between SIP clients and also I tested the SER with the SIP clients which are behind NAT demonstrate its NAT module works fine. Here is my current setup parameter to access landline call: * SER IP address: 207.34.42.53 * SER domain: ubuntu-serv.emerj.net * SIP gateway: sip.broadvoice.com * Internal user ID: 10002 * BroadVoice contact: 7184122016 * Number wants to access through SER: 9178462343 What I have in my MySQL server is as follow: * Domain ubuntu-serv.emerj.net in table 'domain'. * User 10002 registered with domain ubuntu-serv.emerj.net in table 'subscriber'. The password for this account is the normal password I set for it in database. * User 7184122016 registered with domain sip.broadvoice.com in table 'subscriber'. The password for this account is the password has been given by BroadVoice to us. * Defined a record in table 'usr_preferences' contains the following information: +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ | | 9178462343 | | callfwd | sip:7184122016@sip.broadvoice.com | 0 | 2008-08-11 10:00:29 | +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ This is the information I am using for my call forwarding purpose. Whenever the user tries to dial 9178462343, this record blindly forward it to the BroadVoice.com. Here are my questions: 1. I monitored and captured the packet between SIP, SER and BroadVoice. For the normal call directly from SIP to BroadVoice and also between SIP client and SER, a kind of SIP REGISTER message is sent at initial time and it seems that all the rest messages is based on the session value that is initially exchanged. When I try to use my SER as a call forwarding server, and when my SIP client tries to dial, SER doesn't send REGISTER message for account to 7184122016 to the BroadVoice. I want to know whether it is possible for SER to do this REGISTER before sending INVITE message for 9178462343 or not. 1. Is the call forwarding is the correct solution for this issue or not? Maybe there are some other things that I am not aware of them due to I am really new to SER environment. 1. I also saw that SER doesn't overwrite the values of Contact, To, and From in the INVITE message properly and everything is sent to the BroadVoice with the values of the ubuntu-serv.emerj.net as a domain and also initial SIP contact that all of them are unknown for BroadVoice gateway. How should I tell SER to overwrite this values and keeps the track of the changes? I also attached my both ser.cfg and the logs of the SER with my email. I would be really appreciated if you assist me resolving this issue and helping me with any document or sample ser.cfg file showing me how to realize the idea. I am looking forward to you response. --- Sincerely yours, Morteza Alizadeh Design Engineer Cell Phone: +60 (12)-319-7439 Email: malizadeh(a)emerj.com Morteza.Alizadeh(a)gmail.com Skype: mortezaalizadeh Block D-17-6, Gurney Height Condominium, Jalan Bukit Keramat, 54000 Kuala Lumpur, Malaysia

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[Kamailio-Users] A function to add uri/header parameters
by Iñaki Baz Castillo 11 Aug '08

11 Aug '08

Hi, for now, the "normal" way to detect if an in-dialog request/response beyonds to a natted dialog is by adding a uri parameter ("nat=yes") in the "Contact" header of the first request and response and macthing that parameter in the RURI of subsequent in-dialog requests. Typically this is done in this way: search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); Well, this mechanims (that is part of official OpenSer courses) is very RCF3261 non compliant: 1) In SIP a header field name is case insensitive but the above function wouldn't match a "CONTACT" of "contact" header names (even if they are 100% valid). 2) "Contact" header can appear abbreviated as "m" (or "M") but the above doesn't match it. 3) 1.1.1. Known Limitations search ignores folded lines. For example, search("(From|f):.*@foo.bar") doesn't match the following From header field: From: medabeda <sip:medameda@foo.bar>;tag=1234 A solution for points 1 and 2 could be using "subst" with insensitive flag enabled, but it could be very anti-friendly. So I wonder: wouldn't be useful a good funtion to add/read/remove header and uri parameters (in any header or uri) being RFC 3261 compliant (this is: case insensitive and allowing header name abbreviations)? -- Iñaki Baz Castillo

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[Kamailio-Users] ISN routing
by Krunal Patel 11 Aug '08

11 Aug '08

Hi, I would like to implement isn routing in openser. I am new to isn routing. Please help me if anybody of you have idea about isn routing. Thanking in advance - Krunal Patel

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[Kamailio-Users] ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2
by mahesh.peddi＠info-spectrum.com 11 Aug '08

11 Aug '08

Hi All, I am working on Openser 1.3.2 version with no_tls support. When i include "modparam("presence", "db_url","mysql://openser:openserrw@localhost/openser")" for the presence and run openserctl moni i am getting the error as ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. And the errors in /var/log/messages Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:table_version: invalid type (2) or nul (0) version columns for presentity Aug 10 09:37:41 rhel4 openser[5530]: ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2 Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:init_mod: failed to initialize module presence Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:main: error while initializing modules Aug 10 09:37:41 rhel4 openser[5530]: NOTICE:presence:destroy: destroy module ... Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session opened for user root by (uid=0) Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session closed for user root And when i check in the data base its showing the version as mysql> select * from version where table_name="presentity" ; +------------+---------------+ | table_name | table_version | +------------+---------------+ | presentity | 2 | +------------+---------------+ 1 row in set (0.00 sec) Please help me how to resolve this. And as mention in the forums i modify the res.c as below shows even i am facing the same error. --------------------------------------------------------------------------------Index: modules/mysql/res.c===================================================================--- modules/mysql/res.c (revision 4448)+++ modules/mysql/res.c (working copy)@@ -103,7 +103,10 @@ break; default:- RES_TYPES(_r)[i] = DB_STRING;+ LM_WARN("unhandled data type column (%.*s) type id (%d), "+ "use DB_STRING as default\n", RES_NAMES(_r)[col]->len,+ RES_NAMES(_r)[col]->s, fields[col].type);+ RES_TYPES(_r)[col] = DB_STRING; break; } }--------------------------------------------------------------------------------I am enclosing my "openser.cfg" and "openserctlrc"Regards,Mahesh PeddiInfospectrum India Pvt. Ltd.Cell: +91 9765775176IP-Phone Ext. - 764

2 1

[Kamailio-Users] naming of open source projects
by Klaus Darilion 11 Aug '08

11 Aug '08

Hi! I really do not understand why so many people think that the name should include the word "open" or should somehow include parts of the protocol (SIP) or its usage (server, proxy ...). If I take a look at very successful open source projects they do not follow this, e.g. Apache, Eclipse, Pidgin, Wireshark, Asterisk, Tomcat, Boost ... regards Klaus

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Re: [Kamailio-Users] poll: project name
by luzango mfupe 11 Aug '08

11 Aug '08

"Free to Talk" Not bad at all i bet most of us here would prefer an English name!!! Why Kamailio is bad??? relax guys u will be able to pronounce it well soon... My choice iiis numberrr 2. On Mon, Aug 11, 2008 at 3:31 PM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: [Kamailio-Devel] poll: project name (Klaus Darilion) > 2. Re: poll: project name (Yazid Hadj Said) > 3. Re: poll: project name (Ardjan Zwartjes) > 4. Re: [Kamailio-Devel] poll: project name (David Villasmil) > 5. Re: [Kamailio-Devel] poll: project name ( I?aki Baz Castillo ) > 6. Re: poll: project name (Carsten Bock) > 7. Re: poll: project name (Alex Balashov) > 8. Re: poll: project name (Johansson Olle E) > 9. Re: [Kamailio-Devel] poll: project name (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 11 Aug 2008 13:39:29 +0200 > From: Klaus Darilion <klaus.mailinglists(a)pernau.at> > Subject: Re: [Kamailio-Users] [Kamailio-Devel] poll: project name > To: miconda(a)gmail.com > Cc: users <users(a)lists.kamailio.net>, devel <devel(a)lists.kamailio.net> > Message-ID: <48A024F1.5030105(a)pernau.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > speaking as user: > 1) Yes, I love it. Please keep it. > > Daniel-Constantin Mierla schrieb: > > Hello everybody, > > > > some people expressed inconveniences regarding the new name, so we would > > like to get your opinion about, therefore this is a poll that will get > > us to a decision. Here is the question: > > > > Do you like the name "Kamailio"? > > 1) Yes, I love it. Please keep it. > > 2) It is OK. > > 3) I do not mind the name at all > > 4) I do not like it > > 5) I hate the name and we should change it > > > > There is no need of explanations, as it is up to personal taste and > > decision. Just vote by option. > > > > Cheers, > > Daniel > > > > PS. for those interested... > > > > As short history about renaming story. On the 28th of March the board > > was informed about issues with SER part of project name. You can easily > > find who has the trademarks. On the 7th of April the board was informed > > about an agreement for the deadline as of September 30, 2008 to use old > > name. From there on, all old releases, documents, etc. can be referred > > by the old name, but nothing new. Other few discussions about what > > should/could be kept/changed followed and on 12th of May started the > > discussion about the new name with two main direction, something > > similar, to remind about old project or just something completely new, > > to be used as brand name. Couple of proposals and discussions regarding > > possible names followed up to 16th of July when was the deadline and the > > start of vote. Majority was reached on 17th but it was required to have > > opinions from all members of the board, last came from Bogdan, on the > > 28th of July. > > > > All discussions about migration were on the board mailing list. No > > change were done to openser.org configuration, just the new domain with > > kamailio in it was pointed to same server. Every step done was reported > > to board, following a schedule discussed there. Another discussion was > > about making the renaming and 1.4.0 release in same day. Being > > practically impossible to do all the work in one day, it was decided to > > do first the announcement about renaming. That happened on 28th when all > > agreed with the new name. > > > > As you can see, there was no 5 months of renaming duration. There was no > > change in openser.org DNS configuration undertaken with/without the > > knowledge of all board members. The new name domains were reserved when > > majority was achieved, in any case; dns entries for new name were > > pointed to old server shortly after booking the domains. > > > > > > ------------------------------ > > Message: 2 > Date: Mon, 11 Aug 2008 13:52:02 +0200 > From: "Yazid Hadj Said" <yazid.hadjsaid(a)gmail.com> > Subject: Re: [Kamailio-Users] poll: project name > To: users(a)lists.kamailio.net > Message-ID: > <8dfa8c1a0808110452w274cb206te3160cbe4dd8fd2(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > > > >Do you like the name "Kamailio"? > >1) Yes, I love it. Please keep it. > >2) It is OK. > >3) I do not mind the name at all > >4) I do not like it > >5) I hate the name and we should change it > > I do not like it. > I think that at least the word "Open" must be included in the new name > > > Regards > Yazid >

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Re: [Kamailio-Users] simple question: log 180, 200 responses
by Yazid Hadj Said 11 Aug '08

11 Aug '08

Hi Ali, I think that you must call : if(status=="180") { xlog("this is a 180 Ringing reply"); }; if(status=="200") { xlog("this is a 200 OK reply"); }; in the onreply_route, note in tne route function

1 0

[Kamailio-Users] simple question: log 180, 200 responses
by Ali Soltani 11 Aug '08

11 Aug '08

Hi, I am using openSER as a simple sip proxy for calls between two sip end-points. By snooping the traffic I can verify that the openSER proxies ALL sip messages/responses but non of the following log statements do not log the response messages (180, 200). Am I missing anything. route { xlog("(route) usr=$ru method=$rm request_port=$rp received_port=$Rp original port=$op destination_port=$dp src_port=$sp"); xlog("(route) reply_status=$rs"); if(status=="180") { xlog("this is a 180 Ringing reply"); }; if(status=="200") { xlog("this is a 200 OK reply"); }; ..... } Thanks,

2 1

Re: [Kamailio-Users] poll: project name
by Yazid Hadj Said 11 Aug '08

11 Aug '08

Hi all, >Do you like the name "Kamailio"? >1) Yes, I love it. Please keep it. >2) It is OK. >3) I do not mind the name at all >4) I do not like it >5) I hate the name and we should change it I do not like it. I think that at least the word "Open" must be included in the new name Regards Yazid

1 0

[Serusers] 404 Not Found
by dwianto dana 11 Aug '08

11 Aug '08

Hi, I tried to use SER openimscore and connect it with sailfin. Tried to register using X-Lite. What I get in the Core is SIP: Status:404 Not Found - destination user not found on this S-CSCF I am using a default user (Alice and Bob),, so it is kinda strange to get this information, user not found. Thanks for the answers given

1 0

[Kamailio-Users] Req for Coments about my config
by Ali Jawad 11 Aug '08

11 Aug '08

Dear All I am working with some developers to create a sip client that supports TLS for me. During the beta stage they created the phone and it sent "sips" packets with transport=tls in the packet. The thing is that I used the sethostport and t_relay to send data to my PSTN gateway that works on UDP but in both cases I was not able to make PSTN calls. The developers created a custom openser.cfg that rewrites sips packets to sip and removes the transport=tls from the packets. How efficient is this ? Is this the typical approach to make a TLS call through Openser where the PSTN gateway does not support TLS ? Please have a look at the config file if you can. http://pastebin.com/m20f7d25a Thanks

2 1

Re: [Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008
by Cesar Troncoso 11 Aug '08

11 Aug '08

Hello! I am also up for a meeting in Bucharest on Monday next week. It will be a good chance to get different points of view about the kamailio/opensips discussion and the way developers and board members think to move forward in the future. Regards, Cesar ------------------------------ Message: 6 Date: Tue, 05 Aug 2008 18:57:50 +0300 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008 To: users <users(a)lists.kamailio.org>, devel <devel(a)lists.kamailio.org>, business(a)lists.kamailio.org Message-ID: <4898787E.7020602(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Everybody, as many expressed their intention for an openser/kamailio event in Bucharest, next week, I am going to organize a dinner in a place with nice Romanian beer, wine and food. Seeing the suggestions for the date, I am proposing August 11, 2008 (Monday) -- let me know if you have opinions. Exact hour and place will be sent as soon as I get an idea about number of participants. So far, expressed the intention to participate: - Henning Westerholt, developer and board member - Elena-Ramona Modroiu, developer and board member - Daniel-Constantin Mierla, developer and board member - Michiel ten Hagen - Sebastian Schuman Please write me if you want to attend, I need to know the number of attendants to make the reservation. The event is in the same fashion as the ones happening in Barcelona, Vienna, Berlin, ..., in the past -- everybody pays for its expenses. If you want to have some preamble (might not be same place, but...), you can check some photos from last similar event in Bucharest after the ROSDEV2008 conference: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/index1.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] fr_inv_timer issue
by Kaeman Chris 11 Aug '08

11 Aug '08

Hello, We have an fr_inv_timer set to 45 secs. When i make calls to an 800 number, the call hits to the 1st provider on LCR and IVR recording starts asking to hit an option 1,2,3,etc & if the caller dont hit any option and leave the call up to 45 secs. Than fr_inv_timer kills the transaction by sending a CANCEL message and thereby openser forces the call to send it over to the fail over route. Is this correct should it try the call on the failover route because the 1st provider was able to complete the call than why the call went over to the failover route? here as soon as the call switches to the 2nd provider the IVR message starts again from the option 1 leaving the option 3 in the middle of the recording. is this correct? let me know if the query is not clear. Thanks in advance --------------------------------- Unlimited freedom, unlimited storage. Get it now

3 2

[Kamailio-Users] Radius + openser
by Amit Jain 11 Aug '08

11 Aug '08

Hello everyone, I was looking for documentation for Radius AAA + Openser. I tried to google but the links provided are not working. i think since the site has been renamed. Can any provide me document if saved or any other related link. Link to which i was referring . http://openser.org/docs/openser-radius-1.0.x.html Regards, Amit Jain

2 2

[Kamailio-Users] Radius + openser
by Amit Jain 11 Aug '08

11 Aug '08

Hello everyone, I was looking for documentation for Radius AAA + Openser. I tried to google but the links provided are not working. i think since the site has been renamed. Can any provide me document if saved or any other related link. Link to which i was referring . http://openser.org/docs/openser-radius-1.0.x.html Immediate help required :( Regards, Amit Jain

2 1

[Kamailio-Users] Should really 't_on_branch' remain in serial forking?
by Iñaki Baz Castillo 11 Aug '08

11 Aug '08

Hi, imagine this simple case: -------------------------- route { t_on_branch("1"); t_on_failure("2"); xlog("We are in 'route'"); t_relay("1.1.1.1"); } branch_route[1] { xlog("We are in 'branch_route[1]'"); # do something.... } failure_route[2] { xlog("We are in 'failure_route[2]'"); append_branch(); t_relay("2.2.2.2"); } --------------------------- In case an error occurs when forwarding the request to 1.1.1.1 this would be the screen log output: We are in 'route' We are in 'branch_route[1]' We are in 'failure_route[2]' We are in 'branch_route[1]' This is: branch_route[1] will also be runned **again** after the failure route since it was loaded in the first forward attemp. I really don't know if this is intuitive or not. The only way to "dissable" branch_route[1] in the failure route is by adding: ------------------- failure_route[2] { t_on_branch("2"); # <--- Dissable t_on_branch("1") xlog("We are in 'failure_route[2]'"); append_branch(); t_relay("2.2.2.2"); } branch_route[2] { xlog("We are in 'branch_route[2]'"); # Nothing to do here. # This route is neccesary to dissable t_on_branch("1") } ------------------ This would show: We are in 'route' We are in 'branch_route[1]' We are in 'failure_route[2]' We are in 'branch_route[2]' What about if "t_on_branch" wouldn't remain loaded after a failure route or serial forking? wouldn't be more intuitive to re-enable it explicitely when required? -- Iñaki Baz Castillo

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[Kamailio-Users] Res: serMyAdmin login problem invalid username and/or password
by Flavio Goncalves 10 Aug '08

10 Aug '08

Hi Muhammad, I will check the problem this week and return to you soon. Regards, Flavio ----- Mensagem original ---- De: muhammad akl <muhammad.akl(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: users(a)lists.kamailio.org Enviadas: Domingo, 10 de Agosto de 2008 12:44:47 Assunto: Re: [Kamailio-Users] serMyAdmin login problem invalid username and/or password thx Flavio for ur response but i still have another question the main problem with the user "admin@localhost " and password "openserrw" that were created by the openser_mysql.sh is still happening got the same message "invalid username and/or password " but when i removed that account from the subscriber table and used the openser.sql file from sermyadmin i have successfully logged in but after modifying the openser.sql file to match my environment so after that i tried to register with a new account from the home page using register button besides to login button and filled the account's information there everything worked fine but the problem happend here now is when i logged with the "admin@setup" and password "secret " account to approve the new account for users i got this message pastebin.com/m7154020b p.s for alot of testing i have used more than one version of serMyAdmin beginning with ver 0.3 to 0.7 !! sorry for my poor english as am not native BR muhammad

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[Kamailio-Users] Res: serMyAdmin login problem invalid username and/or password
by Flavio Goncalves 10 Aug '08

10 Aug '08

Hi Muhammad, Check the updated procedure at www.sermyadmin.org. The default login is asmin@setup, password secret as stated in the procedure. Flavio E. Goncalves ----- Mensagem original ---- De: muhammad akl <muhammad.akl(a)gmail.com> Para: users(a)lists.kamailio.org Enviadas: Domingo, 10 de Agosto de 2008 6:28:53 Assunto: [Kamailio-Users] serMyAdmin login problem invalid username and/or password hey all i'm trying to configure sermyadmin web interface to administrate openser-1.2.2 through it firstly i installed apache-tomcat and configure it successfully then i downloaded the sermyadmin.war and copied it to /usr/local/tomcat/webapps then downloaded mysql-jdbc and copied it to /usr/local/tomcat/lib also configure the context.xml and modified the Context element and added the user have full privileges to access openser databases then when tried to login with user "admin@locahost" and pass " openserrw" default values created by the openser_mysql.sh i got invalid username or password ? any suggestions ? serMyAdmin version used .7-BETA i also tried the howto found in sermyadmin.org but with no success by trying to use the prepared openser.sql file if anyone could i'd appreciate that

2 1

[Kamailio-Users] How to revert changes in SDP during serial forking?
by Iñaki Baz Castillo 10 Aug '08

10 Aug '08

Hi, I have a scenario in which the call is first forwarded to user location using MediaProxy (so I modify the SDP in OpenSer) and if it gets a error response a failure route forwards the request to a media server with public IP for which I don't need and don't want to use MediaProxy. But since I change the SDP in the first forwarding (to the user) then I can't use the original SDP in the sequential forwarding to the media server. Is possible to do it? Thanks a lot. -- Iñaki Baz Castillo

2 2

[Kamailio-Users] serMyAdmin login problem invalid username and/or password
by muhammad akl 10 Aug '08

10 Aug '08

hey all i'm trying to configure sermyadmin web interface to administrate openser-1.2.2 through it firstly i installed apache-tomcat and configure it successfully then i downloaded the sermyadmin.war and copied it to /usr/local/tomcat/webapps then downloaded mysql-jdbc and copied it to /usr/local/tomcat/lib also configure the context.xml and modified the Context element and added the user have full privileges to access openser databases then when tried to login with user "admin@locahost" and pass " openserrw" default values created by the openser_mysql.sh i got invalid username or password ? any suggestions ? serMyAdmin version used .7-BETA i also tried the howto found in sermyadmin.org but with no success by trying to use the prepared openser.sql file if anyone could i'd appreciate that

1 0

[Kamailio-Users] OpenSIPS project
by Bogdan-Andrei Iancu 10 Aug '08

10 Aug '08

Hello everybody, A new project - OpenSIPs (Open SIP Server) - became active today, with its first release (OpenSIPS 1.4.0). What is OpenSIPS about? OpenSER History: ---------------- Voice System started the OpenSER project with a vision of running an open project that will deliver a professional software. Along the years Voice System was the main contributor and sustainer of the OpenSER project, organizing conferences, events and tutorials, all aiming to consolidate OpenSER as a top solution for industry VoIP. OpenSER Issues: --------------- In the past time, several critical problems related to OpenSER project were identified by us: - the need for a new reliable release - delayed or low-quality releases are affecting the credibility of the project - degradation of the project quality - testing, performance measurements and code quality control are overlooked - lack of the control, management and coordination of the project - critical project issues could not be handled or solved OpenSIPS Solutions: -------------------- OpenSIPS is a continuation of the OpenSER project - we have a moral obligation to develop and deliver the high quality and reliable software we envisioned when starting OpenSER. 1) a 6 months release cycle, but with no compromise for the technical issues 2) extensive testing and performance measurements before each major release 3) contributions are guaranteed to be answered and integrated (if accepted) in less than 1 month 4) code review, architecture control and integrated design for software quality 5) technical management with a strict developer hierarchy and developer cooperation to prevent deadlocks or un-availabilities 6) intellectual honesty by having decisions taken on value-based criteria and arguments. To find more about the project, see the official web page : http://www.opensips.org OpenSIPS 1.4.0 release: ----------------------- First major release of OpenSIPS is n**ow available. It incorporates new functionalities like * dialog profiling extension * local route for internally generated requests * dial plan / translation module * general NAT traversal module (for non-invite signalling) * peering and rate limit (traffic shaping) modules , enhancements like: * SRV load balancing (based on weights) * nonce re-usage verification (safer authentication) and bug fixing - code, specs and functionality bugs. The full Changelog is available on the web site: To get the OpenSIPS 1.4.0 version, see the download page at: http://www.opensips.org/index.php?n=Resources.Downloads Best regards, Bogdan

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[Kamailio-Users] 603 Declined
by Kaeman Chris 10 Aug '08

10 Aug '08

Hello, Does openser as soon as receives 603 Declined from the termination kills the call and not route it to the failover route? Is this correct? Thanks, KC --------------------------------- Unlimited freedom, unlimited storage. Get it now

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[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 09 Aug '08

09 Aug '08

I will be out of the office starting 08/08/2008 and will not return until 08/11/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

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[Kamailio-Users] Compiling MediaProxy 2.0 on CentOS 5
by Andrew Yager 09 Aug '08

09 Aug '08

Hi, Apologies if you have already seen this on the OpenSIPS users list; it seems that some of the domain changes are a little confusing for the moment. I'm trying to compile MediaProxy 2.0 on CentOS 5 for use with Kamailio. I've installed libnetfilter_conntrack from RPMForge, but I am getting an error with compiling. ./setup.py build running build running build_py running build_ext building 'mediaproxy.interfaces.system._conntrack' extension gcc -pthread -fno-strict-aliasing -DNDEBUG -O2 -g -pipe -Wall -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -m64 -mtune=generic -D_GNU_SOURCE -fPIC -fPIC - DMODULE_VERSION=2.0.3 -DIPTABLES_VERSION="1.4.0" -Imediaproxy/ interfaces/system/libiptc/include -I/usr/include/python2.4 -c mediaproxy/interfaces/system/_conntrack.c -o build/temp.linux- x86_64-2.4/mediaproxy/interfaces/system/_conntrack.o mediaproxy/interfaces/system/_conntrack.c: In function ‘ForwardingRule_init’: mediaproxy/interfaces/system/_conntrack.c:223: error: ‘NFCT_SOPT_SETUP_REPLY’ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:223: error: (Each undeclared identifier is reported only once mediaproxy/interfaces/system/_conntrack.c:223: error: for each function it appears in.) error: command 'gcc' failed with exit status 1 Current IPTABLES info: % iptables --version iptables v1.3.5 % lsmod | grep conntrack ip_conntrack_netbios_ns 36033 0 ip_conntrack 91237 1 ip_conntrack_netbios_ns nfnetlink 40457 1 ip_conntrack Is there a package I am missing, or am I simply fighting a loosing battle and need to use a different flavor of Linux to get it to work easily? Thanks, Andrew -- Andrew Yager, Managing Director (MACS BCompSc MCP) Real World Technology Solutions Pty Ltd ph: 1300 798 718 or (02) 9037 0500 fax: (02) 9037 0591 mob: 0405 152 568 http://www.rwts.com.au/ or http://www.stonebridgecomputing.com.au/

1 0

[Kamailio-Users] failure route issue
by Mouncif Benniane 08 Aug '08

08 Aug '08

I am using openser 1.3, I can't make the failure_route to work properly. here is my config: modparam("tm", "fr_timer", 2) # set fr_timer route{ if (is_method("INVITE")) { avp_delete("*"); if (avp_db_query("SELECT prefix,grp_id,priority,custid,did,ip,port,failure_set,failure_i p,failure_port FROM test_table WHERE prefix='$rU'","$avp(s:prefix);$avp(s:grp_id);$avp(s:priority);$avp(s:custid);$avp(s:did);$avp(s:ip);$avp(s:port );$avp(s:failure_set);$avp(s:failure_ip);$avp(s:failure_ port)")) { $ru = "sip:"+$rU + "@" + $avp(s:ip) + ":" + $avp(s:port); if ($avp(s:failure_set) == y ) { $avp(i:3) = 3; t_on_failure("1"); t_relay(); } } } else { xlog("L_NOTICE", "avp_db_query() returned no records or failed query\n"); }; } ### END OF ROUTING failure_route[1] { # status is 408 if openser session timer fires if (t_check_status("408")) { $ru = "sip:"+$rU + "@" + $avp(s:failure_ip) + ":" + $avp(s:failure_port); t_relay(); } } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } So I set ip field to no responsive one and failure_ip to responsive one, but still the invite does not fail to the last one (failure_ip) after the timeout. Am I doing something wrong? cheers!!

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[Kamailio-Users] Kamailio in orange (logo submission).
by Alex Balashov 08 Aug '08

08 Aug '08

I have a logo submission from someone who works with me at Evariste - of artistic inclinations. I don't know what the proper submission path for the logos is, so I'm posting it here. Sorry if that's annoying. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] parameter 'ftag' and 'avp'
by Steven Wu 08 Aug '08

08 Aug '08

Hi all, What is meaning of ftag or avp parameter in Record-Route herder? SER inserted this Record-Route. As far as know, OpenSER is come from SER. Best regards, Steven Wu

2 1

[Kamailio-Users] comments to pointed openser issues
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello, these issues were pointed in the first place, my comments follow inline. 1) the need for a new reliable release - delayed or low-quality releases are affecting the credibility of the project - http://lists.kamailio.org/pipermail/users/2008-August/018999.html - the link above shows this one was a faster release than in past - http://www.kamailio.net/mos/view/KAMAILIO-v1.4.0-Released/ - the link above shows the efforts in creating a testing suite which was running on devel server during this development cycle - the link above shows no much new things in the core, but code cleanup, duplicity removal, source code documentation 2) degradation of the project quality - testing, performance measurements and code quality control are overlooked - presented before, testing suite was built and running, lot of code was reviewed and improved - who in the new project will do that and couldn't do it because was blocked in the old project 3) lack of the control, management and coordination of the project - critical project issues could not be handled or solved - I am not aware of mis-coordinations that could affect the project so badly - I am not aware of critical issues that could not be solved and there was no report of a critical deadlock - if there was a failure in management, all from the board share same fault Moreover, I believe there are facts in the past that openser took care of its quality. I repeat as I said that QA might not be enough no matter is done. If openser 1.3.x is the core component of a platform hosting about 2 millions of subscribers does not ensure some QA, then nothing else can be said, see: http://www.ilocus.com/2008/03/the_largest_voip_offering_base.html None above motivates such reactions as happened recently, but if does, then probably with some openess and discussions everything would have been better now. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] OpenSIPS vs. Kamailio
by Darren Sessions 08 Aug '08

08 Aug '08

I have to say, this is probably one of the harder decisions I've had to make in a very long time. I have to say the OpenSER / SER decision was much easier. While I obviously don't know all the details of what's gone on between Bogdan and Henning or whomever else, I just really have a hard time swallowing Hennings explanation of what happened and his, what seems to be, emotional reaction. More so than that, it seems like there were quite a few decisions made by the board without immediately letting the community know about it, even quite a while after these decisions were made. While I do not dispute Hennings technical ability, It is my opinion that his disrespectful attitude, his "shotgun" conduct, and his obvious lack of management ability is not befitting of someone on Kamailio project's board regardless of who is at fault here. Honestly, I'm a bit disappointed that Daniel hasn't said more as well. I met him at the fall VON in 2005 and he was indeed very pleasant and I enjoyed our conversation. All I've heard is things like 'not wanting to discuss any further', etc. - but the fact remains that wether it's a comfortable topic or not, there are quite a few of us that are using OpenSER in a commercial environment and these are things that we NEED to know in order to make an informed decision. In the end, all I know is now there are two projects and I have to choose which one. Based on what I've read in the change-logs, my direct experience with Bogden in a technical support role directly related to SER at a previous business - my company will be using OpenSIPS going forward. I have a very high regard for virtually all of the Kamailio / OpenSER group, and again, I am disappointed and absolutely hate that I've been forced into a position to have to make a decision like this. I will continue to be active on both sets of lists for the foreseeable future to monitor progress. Having said all that, I can't tell you how much I hope things can still be resolved and we can all have a unified project. I would implore the Kamailio board to do whatever it takes to bring Bogdan back into the group for the sake of the community. Hopefully something like that won't be just a dream. Thanks, - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com http://www.linkedin.com/in/dsessions _____________________________

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[Kamailio-Users] releases history
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello, as there were accuses about a delayed release, I am showing the dates when a major release was out: June 14, 2005 - first openser release, just a rebuild of ser 0.9.x October 28, 2005 - openser 1.0.0 July 10, 2006 - openser 1.1.0 (after more than 8 months) March 12, 2007 - openser 1.2.0 (after more than 8 months) December 13, 2007 - openser 1.3.0 (after more than 9 months) August 7, 2008 - openser 1.4.0 (less than 8 months) So, maybe my memory is damaged, but you should be able to learn it from mailing lists and web. I do not buy that motivation about project quality degradation in release management. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] openser board summary
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello guys, first I apologize for the situation we are in and for my silence in the last days as I tried to get the release out and not put gas on fire. The board looks like being now an evil thing, maybe it is our fault we haven't given more details about insights, but there were not many as ones could imagine and the decisions were presented to community. Here are the outlines of the discussions on the board, since September 2005, when first mail came there: 2005-09 DCM: decide release number => resulted in openser 1.0.0 DCM: business mailing lists => resulted in creation of the list 2005-10 AG: asks about making AG a trusted CA for default delivery of openser => self signed or something invalid to force users to change 2005-11 DCM: starts discussion about making an openser statute DCM: puts in dicussion the behavior of Harry on the mailing lists DCM: contacts Harry 2005-12 ALL: continuation of previous mail discussions 2006-03 DCM: plans for new release 2006-04 JH: discussion about presence implementation 2006-05 BIA: about project hosting: still sourceforge or move out ALL: Sourceforge problems discussions JH: presented some options for hosting 2006-06 AG: sip summer code JH: about 1.1.0 2006-07 JH: presence server AG: minutes if SIMPLE WG at IETF DCM: GPL issue with openssl and debian 2006-08 BIA: openser summit 2006-09,10 ALL: openser summit 2006-11 AG: history update in relation with old project BIA: TLS discussion 2006-12 ALL: history update 2007-01 KD: FOSDEM DCM: openser 1.1.1 BIA: openser at spring von 2007-02 BIA: openser donnations DCM: openser events 2007-04 BIA: Linuxtag 2007-09 DCM: summit 2007 2007-10 CS: retires from board BIA: proposes Henning => no decision as no rules for electing, makes that a point in todo BIA: openser 1.2.3 2007-11 DCM: openser business environment => hosting it within the website couldn't get a decision, posponed with alternative for dedicated site BIA: new developer Maxim Sobolyev 2008-03 BIA: on the 28th announces to board the issues with trademark, apparently some board member and other people were already aware AG: openser pages update 2008-04 ALL: agreed with rename - started the idea of a foundation to register trademark, takes on managing resources HW (not a member at that time but was included in the discussions related to TM) - proposes new developer Iouri 2008-05 BIA: on the 12th started the discussion about new name foreseeing 2 directions: something similar, to remember about the old project, or something totally different 2008-06 DCM: 9th: paypal report for last year => total donations to project's sourceforge account 52.05Euro (DCM Paypal account closed, the money still at DCM, proposed to use them for buying domains for new name) DCM: 9th: forwards a proposal received at end of April to team(a)openser.org from PacktPub to donate 2% or 5% (if we keep news on the site about the book) of openser book sales to the project. DCM asked for decision as there was no legal entity to deal with. (note: team(a)openser.org was created at the time openser forked in 2005, to get private feedback at that time, was used for the summit 2006 for contact, open for any post, includes the founders, about 10 good mails out of thousands spams per year) BIA: 9th: adds that in a discussion with PacktPub (not on team or board, as I haven;t seen such) proposed to give the 5% for openser devel course organized by him. The course was announced to the lists on the 4th of June. DCM: 9th: expressed surprise as he thought that was a different deal, made privately between BIA and PacktPub BIA: ask the board about donation, saying that actually the 5% donation was not really accounted ALL: There was no decision of what to do with the sponsorship as the board has no treasurer and we are working towards creating a foundation that can accept legally fundings. There was no ask for some amount of money for the devel course, no costs presented, no other details given. Nobody expressed a disagreement of giving funds for the devel course. BIA: 9th : replacement for Cesc as quick solution to deal with TM => suggests Henning Westerholt, all the other agree up to 17th DCM: asks Henning and then sends short announcement to community BIA: complains that he actually hasn't voted and the replacement was made in hurry (?!?!) DCM: presents that all other announcements were done more or less in the same fashion in the past, asking to propose improvements DCM: 24th - asks decision about local_route, board discusses and agrees to enclose in #ifdef HW: 25th - announces the result and asks BIA if he agrees, no response, he was away HW: after 1 day since asking BIA for agreement SVN is updated enclosing local_route in ifdefs BIA: 27th - disagrees with board decision, considering it was not the business of the board (sent to public as well) -- Note from DCM: opensips was registered at June 25, 2008, 17:30 according to http://sourceforge.net/projects/opensips/ Discussions with name proposals and foundation 2008-07 Others: discussions related to name and foundation HW: proposes a project renaming plan HW: on the 16th, vote for name: proposes Kamailio, majority reached on the 17th, last that agree (with a mention that he does not really like the name) BIA, on the 28th, after a call from HW where he mentioned he was away. AG: 17th - registers kamailio.org, ... AG: prposes unification of 1.4. and renaming => decided to announce first to allow time for renaming without pressure AG: 28th - announces renaming to public HW: some drafts of foundation if in Germany AG: details about alternative in Netherlands 2008-08 DCM: 4th - discovers news on voip-info.org that openser becomes opensips, Dan Pascu sends a mixed recipient message and some others see something is going on DCM: after short investigation and learning that behind opensips is bogdan decide to take some precaution actions and send details about them on board to decide if they still remain or roll back BIA: comes later on the mailing lists and announces the fork From then on most of the things were reflected on the public. No one was kicked out from the project, as developer, board member or simply list user, just some administrative rights were limited. Of course last three months have a lot of messages for the topics above. I hope I haven't missed something important and I ask other board members to correct or improve. With this one I hope things are a bit more clear about what board did so far, where were mistakes and wrong decisions. You judge! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] ser 2.1 - compilation error - conflicting types
by Tomasz Zieleniewski 08 Aug '08

08 Aug '08

Hi, When I try to compile ser from sources I get the following conflicting types errors: gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.1"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lock_ops.c -o lock_ops.o In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:46: error: conflicting types for 'loff_t' /usr/include/linux/types.h:30: error: previous declaration of 'loff_t' was here /usr/include/sys/types.h:62: error: conflicting types for 'dev_t' /usr/include/linux/types.h:13: error: previous declaration of 'dev_t' was here In file included from /usr/include/sys/types.h:133, from ut.h:53, from lock_ops.c:28: /usr/include/time.h:105: error: conflicting types for 'timer_t' /usr/include/linux/types.h:22: error: previous declaration of 'timer_t' was here In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:198: error: conflicting types for 'int64_t' /usr/include/linux/types.h:98: error: previous declaration of 'int64_t' was here /usr/include/sys/types.h:204: error: conflicting types for 'u_int64_t' /usr/include/linux/types.h:97: error: previous declaration of 'u_int64_t' was here In file included from /usr/include/sys/types.h:220, from ut.h:53, from lock_ops.c:28: /usr/include/sys/select.h:78: error: conflicting types for 'fd_set' /usr/include/linux/types.h:12: error: previous declaration of 'fd_set' was here In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:235: error: conflicting types for 'blkcnt_t' /usr/include/linux/types.h:124: error: previous declaration of 'blkcnt_t' was here make: *** [lock_ops.o] Błąd 1 Is there any fix for that? Regards Tomasz

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[Kamailio-Users] mailing lists issues
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

Hello, I am not going in hot discussions here. We have to finish the release first. Due to DNS change of openser.org, without prior notice, we were facing about 1 hour down time with mailing lists. I apologize for that. Kamailio project should be reachable on two domains: - main one: kamailio.org (www.kamailio.org) - backup: kamailio.net (www.kamailio.net) The mailing lists domains are: - main one: lists.kamailio.org - backup: lists.kamailio.net If you have any problem, please contact me. Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] carrierroute compile error
by Mouncif Benniane 07 Aug '08

07 Aug '08

That fixed it thanks On Thu, Aug 7, 2008 at 4:03 PM, TCB <tawandac(a)gmail.com> wrote: > you need libconfuse-deve ! > > > On Thu, Aug 7, 2008 at 9:38 PM, Mouncif Benniane <mouncifbb(a)gmail.com>wrote: > >> I did install libconfuse rpm for centos 5 but still same compile results >> >> [root@r sip-server]# rpm -qa | grep confuse >> *libconfuse-2.5-4.el5* >> [root@r sip-server]# >> >> On Thu, Aug 7, 2008 at 3:30 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> You need libconfuse, don't know exactly the name of the package on >>> centos. >>> >>> http://www.kamailio.net/docs/modules/1.3.x/carrierroute.html#AEN53 >>> >>> Cheers, >>> Daniel >>> >>> PS. Please note that the new mailing list address is >>> users(a)lists.kamailio.net >>> >>> >>> On 08/07/08 22:25, Mouncif Benniane wrote: >>> >>>> I am compiling version openser 1.3.2 on centos realease 5 >>>> >>>> getting compile error stooping at carrierroute module see below: >>>> >>>> >>>> make[1]: Entering directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> make[1]: Leaving directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> make[1]: Entering directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> Compiling carrierroute.c >>>> Compiling carrier_tree.c >>>> Compiling load_data.c >>>> Compiling route_config.c >>>> route_config.c:39:21: error: confuse.h: No such file or directory >>>> route_config.c:50: error: expected declaration specifiers or â...â >>>> before âFILEâ >>>> route_config.c:52: error: expected â=â, â,â, â;â, âasmâ or >>>> â__attribute__â before â*â token >>>> route_config.c:63: error: expected â)â before â*â token >>>> route_config.c: In function âload_configâ: >>>> route_config.c:81: error: âcfg_tâ undeclared (first use in this >>>> function) >>>> route_config.c:81: error: (Each undeclared identifier is reported only >>>> once >>>> route_config.c:81: error: for each function it appears in.) >>>> route_config.c:81: error: âcfgâ undeclared (first use in this function) >>>> route_config.c:83: error: âdâ undeclared (first use in this function) >>>> route_config.c:83: error: âpâ undeclared (first use in this function) >>>> route_config.c:83: warning: left-hand operand of comma expression has no >>>> effect >>>> route_config.c:83: error: âtâ undeclared (first use in this function) >>>> route_config.c:83: warning: left-hand operand of comma expression has no >>>> effect >>>> route_config.c:93: warning: implicit declaration of function >>>> âparse_configâ >>>> route_config.c:104: warning: implicit declaration of function âcfg_sizeâ >>>> route_config.c:112: warning: implicit declaration of function >>>> âcfg_getnsecâ >>>> route_config.c:113: warning: implicit declaration of function >>>> âcfg_titleâ >>>> route_config.c:113: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:118: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:123: warning: implicit declaration of function >>>> âcfg_getintâ >>>> route_config.c:127: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:132: warning: implicit declaration of function >>>> âcfg_getfloatâ >>>> route_config.c:134: warning: implicit declaration of function >>>> âcfg_getstrâ >>>> route_config.c:134: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:135: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:137: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:145: warning: implicit declaration of function >>>> âcfg_getnintâ >>>> route_config.c:169: warning: implicit declaration of function âcfg_freeâ >>>> route_config.c: At top level: >>>> route_config.c:178: error: expected â=â, â,â, â;â, âasmâ or >>>> â__attribute__â before â*â token >>>> route_config.c: In function âsave_configâ: >>>> route_config.c:234: error: âFILEâ undeclared (first use in this >>>> function) >>>> route_config.c:234: error: âoutfileâ undeclared (first use in this >>>> function) >>>> route_config.c:241: warning: implicit declaration of function âfopenâ >>>> route_config.c:249: warning: implicit declaration of function âfprintfâ >>>> route_config.c:249: warning: incompatible implicit declaration of >>>> built-in function âfprintfâ >>>> route_config.c:251: error: too many arguments to function >>>> âsave_route_data_recursorâ >>>> route_config.c:257: warning: implicit declaration of function âfcloseâ >>>> route_config.c: At top level: >>>> route_config.c:274: error: expected declaration specifiers or â...â >>>> before âFILEâ >>>> route_config.c: In function âsave_route_data_recursorâ: >>>> route_config.c:280: warning: incompatible implicit declaration of >>>> built-in function âfprintfâ >>>> route_config.c:280: error: âoutfileâ undeclared (first use in this >>>> function) >>>> route_config.c:326: error: too many arguments to function >>>> âsave_route_data_recursorâ >>>> route_config.c: In function âbackup_configâ: >>>> route_config.c:335: error: âFILEâ undeclared (first use in this >>>> function) >>>> route_config.c:335: error: âfromâ undeclared (first use in this >>>> function) >>>> route_config.c:335: error: âtoâ undeclared (first use in this function) >>>> route_config.c:335: warning: left-hand operand of comma expression has >>>> no effect >>>> route_config.c:364: warning: implicit declaration of function âfeofâ >>>> route_config.c:365: warning: implicit declaration of function âfgetcâ >>>> route_config.c:366: warning: implicit declaration of function âferrorâ >>>> route_config.c:370: warning: implicit declaration of function âfputcâ >>>> route_config.c:377: error: âEOFâ undeclared (first use in this function) >>>> make[1]: *** [route_config.o] Error 1 >>>> make[1]: Leaving directory >>>> `/usr/local/src/openser-1.3.0/sip-server/modules/carrierroute' >>>> make: *** [modules] Error 2 >>>> >>>> >>>> >>>> >>>> any ideas? thank you >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierla >>> http://www.asipto.com >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > TC >

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[Kamailio-Users] Accounting / REFER question
by Zahid Mehmood 07 Aug '08

07 Aug '08

Hi All, I have a need to keep the openser 'IN' the loop for all transfered calls and to be able to generate proper accounting records (to mimic legacy PBX behavior). Any thoughts on how this can be achieved WITHOUT using SBC/ B2BUA? is it even possible? If off-system user calls X on openser and then X transfers the call to Y, we need to be able to generate two cdr records: A --> X start when A called, end when Y disconnected X --> Y start when X called Y, end when Y disconnected. A is connecting to X via a cisco gateway. Y can potentially be another off-system number. Thanks in Advance. -- Zahid

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[Kamailio-Users] carrierroute compile error
by Mouncif Benniane 07 Aug '08

07 Aug '08

I am compiling version openser 1.3.2 on centos realease 5 getting compile error stooping at carrierroute module see below: make[1]: Entering directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' make[1]: Leaving directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' make[1]: Entering directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' Compiling carrierroute.c Compiling carrier_tree.c Compiling load_data.c Compiling route_config.c route_config.c:39:21: error: confuse.h: No such file or directory route_config.c:50: error: expected declaration specifiers or â...â before âFILEâ route_config.c:52: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token route_config.c:63: error: expected â)â before â*â token route_config.c: In function âload_configâ: route_config.c:81: error: âcfg_tâ undeclared (first use in this function) route_config.c:81: error: (Each undeclared identifier is reported only once route_config.c:81: error: for each function it appears in.) route_config.c:81: error: âcfgâ undeclared (first use in this function) route_config.c:83: error: âdâ undeclared (first use in this function) route_config.c:83: error: âpâ undeclared (first use in this function) route_config.c:83: warning: left-hand operand of comma expression has no effect route_config.c:83: error: âtâ undeclared (first use in this function) route_config.c:83: warning: left-hand operand of comma expression has no effect route_config.c:93: warning: implicit declaration of function âparse_configâ route_config.c:104: warning: implicit declaration of function âcfg_sizeâ route_config.c:112: warning: implicit declaration of function âcfg_getnsecâ route_config.c:113: warning: implicit declaration of function âcfg_titleâ route_config.c:113: warning: assignment makes pointer from integer without a cast route_config.c:118: warning: assignment makes pointer from integer without a cast route_config.c:123: warning: implicit declaration of function âcfg_getintâ route_config.c:127: warning: assignment makes pointer from integer without a cast route_config.c:132: warning: implicit declaration of function âcfg_getfloatâ route_config.c:134: warning: implicit declaration of function âcfg_getstrâ route_config.c:134: warning: assignment makes pointer from integer without a cast route_config.c:135: warning: assignment makes pointer from integer without a cast route_config.c:137: warning: assignment makes pointer from integer without a cast route_config.c:145: warning: implicit declaration of function âcfg_getnintâ route_config.c:169: warning: implicit declaration of function âcfg_freeâ route_config.c: At top level: route_config.c:178: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token route_config.c: In function âsave_configâ: route_config.c:234: error: âFILEâ undeclared (first use in this function) route_config.c:234: error: âoutfileâ undeclared (first use in this function) route_config.c:241: warning: implicit declaration of function âfopenâ route_config.c:249: warning: implicit declaration of function âfprintfâ route_config.c:249: warning: incompatible implicit declaration of built-in function âfprintfâ route_config.c:251: error: too many arguments to function âsave_route_data_recursorâ route_config.c:257: warning: implicit declaration of function âfcloseâ route_config.c: At top level: route_config.c:274: error: expected declaration specifiers or â...â before âFILEâ route_config.c: In function âsave_route_data_recursorâ: route_config.c:280: warning: incompatible implicit declaration of built-in function âfprintfâ route_config.c:280: error: âoutfileâ undeclared (first use in this function) route_config.c:326: error: too many arguments to function âsave_route_data_recursorâ route_config.c: In function âbackup_configâ: route_config.c:335: error: âFILEâ undeclared (first use in this function) route_config.c:335: error: âfromâ undeclared (first use in this function) route_config.c:335: error: âtoâ undeclared (first use in this function) route_config.c:335: warning: left-hand operand of comma expression has no effect route_config.c:364: warning: implicit declaration of function âfeofâ route_config.c:365: warning: implicit declaration of function âfgetcâ route_config.c:366: warning: implicit declaration of function âferrorâ route_config.c:370: warning: implicit declaration of function âfputcâ route_config.c:377: error: âEOFâ undeclared (first use in this function) make[1]: *** [route_config.o] Error 1 make[1]: Leaving directory `/usr/local/src/openser-1.3.0/sip-server/modules/carrierroute' make: *** [modules] Error 2 any ideas? thank you

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[Kamailio-Users] test
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

please ignore -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] SIP 304 Temporary moved: sendto returning errno 1
by Vincent Pelletier 07 Aug '08

07 Aug '08

Hi. I built & installed ser-2.0.0-rc1 on a debian stable amd64. When I try to redirect a call (case where destination user redirects before answering), it gets an SIP error 477. Corresponding error from SER process running in foreground: 0(30330) ERROR: udp_send: sendto(sock,0x2aebd16fc038,1227,0,xx.xx.xx.xx:xxxx,16): Operation not permitted(1) 0(30330) msg_send: ERROR: udp_send failed 0(30330) ERROR: t_send_branch: sending request on branch 0 failed 0(30330) ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Unfortunately, "1" (EPERM) is not documented in sento manpage, as far as I can see, so it's quite hard to track down that problem. Kernel version is: 2.6.24.5-grsec iptables is used, but putting netcat in place of SER and the client shows that kernel allows packets to reach client (and more simply, it is possible to do a direct call, it's just redirects which fail). It can be a grsec problem, I still have to check that part, but I wanted to take the chance of helpful feedback in the meantime. Regards, -- Vincent Pelletier

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Re: [Kamailio-Users] www.openser.org
by Henning Westerholt 07 Aug '08

07 Aug '08

On Thursday 07 August 2008, Nuno Marques wrote: > Can someone tell me how to access the dokuwiki page? Is offline in both > projects (Kamailio and Opensips...). Hi Nuno, there is a wrong link in the page, should be fixed now. Cheers, Henning

1 0

[Kamailio-Users] www.openser.org
by Bogdan-Andrei Iancu 07 Aug '08

07 Aug '08

Staring from today, www.openser.org will point to www.opensips.org . The mainlists will continue on openser like they use to, but I invite everyone to move to the new mail lists created on opensips.org 1 - It is the time to move forward with the openser project, in a positive way, for a reliable future development and enhancement. 2 - Kamillio community used the access right they had on the www.openser.org server to redirect to kamillio without asking my consent even if the domain is a property of Voice System. Thanks, Bogdan

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[Kamailio-Users] [Fwd: [Kamailio-Devel] branching 1.4.0 in one hour]
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

I think this is also useful info for user community as well. Cheers, Daniel -------- Original Message -------- Subject: [Kamailio-Devel] branching 1.4.0 in one hour Date: Thu, 07 Aug 2008 18:09:35 +0300 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: devel <devel(a)lists.kamailio.org> Hello, this is a last notification between branching the 1.4.0 release. If any developers is still doing last minute work, let us know. Branching is about one hour from now 18:00 CET. Of course, will still be possible to add afterwards, just that will need commits in two branches. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Devel mailing list Devel(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Is sh interface is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi, I am working on OpenSER 1.3.2 version as presence server and i want to integrate with IMS. For this what are modifications need. Is sh interface is needed? And how to trace the presence information? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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[Kamailio-Users] REFER & accounting question
by Zahid Mehmood 07 Aug '08

07 Aug '08

Hi All, I have a need to keep the openser 'IN' the loop for all transfered calls and to be able to generate proper accounting records (to mimic legacy PBX behavior). Any thoughts on how this can be achieved WITHOUT using SBC/ B2BUA? is it even possible? If off-system user calls X on openser and then X transfers the call to Y, we need to be able to generate two cdr records: A --> X start when A called, end when Y disconnected X --> Y start when X called Y, end when Y disconnected. A is connecting to X via a cisco gateway. Y can potentially be another off-system number. Thanks in Advance. -- Zahid

1 0

[Kamailio-Users] For xcap-client to make run is xcap-server is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi All, Presently i am working on OpenSER 1.3.2 version to make as presence server and i include the presence module. But i am getting some errors when i explored it seems xcap client was not include.Now i was include but my doubt is for xcap-client to make run is xcap-server is needed. If so where i can get it. Please help me on this. Thanks & Regards, Mahesh Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] For xcap-client to make run is xcap-server is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi All, Presently i am working on OpenSER 1.3.2 version to make as presence server and i include the presence module. But i am getting some errors when i explored it seems xcap client was not include.Now i was include but my doubt is for xcap-client to make run is xcap-server is needed. If so where i can get it. Please help me on this. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] website updates
by Elena-Ramona Modroiu 07 Aug '08

07 Aug '08

Hi all, I have updated the website to reflect the renaming of the project. If you find broken links, please contact me. Best regards, Ramona

1 0

[Kamailio-Users] includes in the route script ?
by Francois Berganz 07 Aug '08

07 Aug '08

is it possible to divide the script route in others files and do includes? With that, I want to have one file for route[1] , one for route[2] .

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Re: [Kamailio-Users] Users Digest, Vol 39, Issue 27
by luzango mfupe 07 Aug '08

07 Aug '08

Hi guys ..tired of this conflict..dont we supposed to have a mojor release today?? On 8/7/08, users-request(a)lists.kamailio.org <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: Question about registrar lookup() and branching. > (Gustavo Mistrinelli) > 2. Re: [Kamailio-Devel] [OpenSER-Users] OpenSER Advanced > Programming Course, Bucharest, Voice System, August 11-14, 2008 > (Andhika Gannesha Gemilang) > 3. Problem using additional attributes through radius_extra > parameter (Juha Heinanen) > 4. Re: [Kamailio-Devel] call for contribution: project logo > (Benny Prijono) > 5. Re: [Kamailio-Devel] call for contribution: project logo > (Torben Friese) > 6. Re: OpenSIPS project (Francois Menard) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 6 Aug 2008 09:59:43 -0300 > From: "Gustavo Mistrinelli" <gmistrinelli(a)gmail.com> > Subject: Re: [Kamailio-Users] Question about registrar lookup() and > branching. > To: "Alex Balashov" <abalashov(a)evaristesys.com> > Cc: users(a)lists.kamailio.org > Message-ID: > <c6717cc80808060559k24a7c774v96049c8fb5c7677a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Alex, > > I don't know your complete architecture, but let me ask you why Asterisk is > not a trusted gateway? > The you can use rewritehost or LCR logic to route DIDs to Asterisk > If you need to associate users with DIDs, you can use dbaliases. > > Cheers, > > Gustavo > On Tue, Aug 5, 2008 at 8:55 PM, Alex Balashov > <abalashov(a)evaristesys.com>wrote: > >> Greetings, >> >> I've got a small problem with how registration lookups work. >> >> I have several concurrent registrations from one PBX that have multiple >> contact URIs that they declare that aren't the same; they contain the >> DIDs to route a call to. So, for instance, in Asterisk: >> >> register => user:pass@registrar/DID1 >> register => user:pass@registrar/DID2 >> >> Of course, lookup() doesn't care what those contacts actually are. If >> it sees multiple contacts for one username, t_relay() will simply create >> branches for all of them and and ring them all. I can't actually have >> that happen. >> >> I'm looking for suggestions on how to handle this most elegantly. The >> two options I've got in mind are: >> >> 1. Stop using lookup() and do my own URI rewrites and branch forking (if >> the contact URIs are identical) with attention to the contact. >> >> 2. Use BRANCH-ROUTEs to arrest the call leg if it is going to an RURI >> user part that does not match the user part of the contact URI, and... >> drop() the branches? >> >> I also need to build in some other logic, such that for example if a >> user has only one contact registered and it does not match the DID, or >> none of the contacts match the DID, proceed as normal anyway. >> >> Is it possible there is a better way? Recommendations welcome. >> >> Cheers, >> >> -- Alex >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

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[Serusers] call via several SER server
by 静于 07 Aug '08

07 Aug '08

hello all, Could two UAs call each other via several SER proxy? I want to implement this function.How to do it?Do I need to modify the ser.cfg? How dose SER server connect each other? --------------------------------- 雅虎邮箱，您的终生邮箱！

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[Kamailio-Users] Problem using additional attributes through radius_extra parameter
by Ruchir 06 Aug '08

06 Aug '08

Hi, I'm trying to configure openser & freeradius with AAA. I'm using custom mysql tables for authentication & accounting. I've modified queries in sql.conf in raddb folder. Authentication works fine. I've added additional fields in radius_extra parameter of acc module. I've added those parameters in dictionaries of radiusclient-ng & freeradius. But I get blank values for custom fields. Any idea what can be wrong? *openser.cfg* modparam("acc", "radius_extra","User-Name=$fU;Dialstatus=$avp(s:dialstatus);User-Rate=$avp(i:2016);User-Duration=$avp(i:2016);User-Clid=$fU;User_ClidName=$avp(s:callerid);User_Dst_Name=$avp(s:2014);User_Dst_Code=$avp(i:2015);User_Account_Id=$fU;Lcr_Rate=$avp(i:2030);Lcr_Duration=$avp(i:2016);Lcrtrunk_Id=$avp(i:2027);Lcr_Dst_Name=$avp(s:2028);Lcr_Dst_Code=$avp(i:2029);Real_Duration=$avp(i:2016)") *dictionary.openser* ATTRIBUTE Dialstatus 1073 string ATTRIBUTE User-Rate 1074 string ATTRIBUTE User-Duration 1075 string ATTRIBUTE User-Channel 1076 string ATTRIBUTE User-Clid 1077 string ATTRIBUTE User_ClidName 1080 string ATTRIBUTE User_Dst_Name 1081 string ATTRIBUTE User_Dst_Code 1082 string ATTRIBUTE User_Account_Id 1083 string ATTRIBUTE Lcr_Rate 1084 string ATTRIBUTE Lcr_Duration 1085 string ATTRIBUTE Lcrtrunk_Id 1086 string ATTRIBUTE Lcr_Dst_Name 1087 string ATTRIBUTE Lcr_Dst_Code 1088 string ATTRIBUTE DialedTime 1089 string ATTRIBUTE Real_Duration 1090 string ATTRIBUTE Startleg 1091 string ATTRIBUTE Stopleg 1092 string *dictionary.radius* ATTRIBUTE Dialstatus 1073 string ATTRIBUTE User-Rate 1074 string ATTRIBUTE User-Duration 1075 string ATTRIBUTE User-Channel 1076 string ATTRIBUTE User-Clid 1077 string ATTRIBUTE User_ClidName 1080 string ATTRIBUTE User_Dst_Name 1081 string ATTRIBUTE User_Dst_Code 1082 string ATTRIBUTE User_Account_Id 1083 string ATTRIBUTE Lcr_Rate 1084 string ATTRIBUTE Lcr_Duration 1085 string ATTRIBUTE Lcrtrunk_Id 1086 string ATTRIBUTE Lcr_Dst_Name 1087 string ATTRIBUTE Lcr_Dst_Code 1088 string ATTRIBUTE DialedTime 1089 string ATTRIBUTE Real_Duration 1090 string ATTRIBUTE Startleg 1091 string ATTRIBUTE Stopleg 1092 string

2 1

[OpenSER-Users] OpenSER Advanced Programming Course, Bucharest, Voice System, August 11-14, 2008
by Bogdan-Andrei Iancu 06 Aug '08

06 Aug '08

Hi everybody, Following the tradition started last year, the second edition of the "OpenSER Advanced Programming Course" will be held in Bucharest, Romania, starting with 11th of August 2008. The primary goal is the same: to attract more developers to OpenSER and build a higher development squad. There are three major upgrades from last edition: - free participation - there will be no fees for attending to this course; you just need to register. - there will be 4 days course instead of 3 days - the course will contain also a practical work where the students will have the chance to consolidate the knowledge they have accumulated during the theoretical presentations. The course will have a 60% - 40% distribution between theory and practice. The course is dedicated to people willing to learn how to develop OpenSER extensions. The course will follow the idea of writing a new openser module from scratch, with all the dependencies it has from the core: dynamic memory usage, MI interface, registering timers, locking, exporting pseudo-variables, etc The course is sponsored courtesy of : - Packt Publishing (donating 5% of the revenue from the Flavio E. Goncalves's "Building Telephony Systems with OpenSER" book.) - Voice System (logistics and hosting) Teachers from the OpenSER development team will be: - Bogdan-Andrei Iancu - Anca Vamanu As, said, the registration is free, but the class is limited as size up to 17 - 20 people. The seats will be granted on the "first in, first served" policy :). For registration, you need to mail me (bogdan(a)voice-system.ro) or to openser(a)voice-system.ro . You will receive a confirmation response to guarantee your registration. Best regards, Bogdan

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[Serusers] Textops problem
by Frank Hase 06 Aug '08

06 Aug '08

Hello, I´m using the SER for my phones at home. Now I want to phone as an anonymous. So I add a line in the ser.cfg: subst('/^From:(.*)sip[^@]*@[a- zA-Z0-9.]+(.*)$From:anonymous@anonymous.invalid;/ig'); But this doesn´t work. I just want to have in the Message Header the line From: anonymous(a)ananymous.invalid; So that I can phone anonym. Thank you for your help. Greeting Frank Haase

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Re: [Serusers] Serusers Digest, Vol 64, Issue 4
by Akintayo Olusegun 06 Aug '08

06 Aug '08

mail it again On Wed, Aug 6, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. 2.0 survey (Jiri Kuthan) > 2. ser 2.1 - compilation error - conflicting types > (Tomasz Zieleniewski) > 3. ser 2.1 - compilation error - conflicting types > (Tomasz Zieleniewski) > 4. SER/FIFO not accessible (cindy.cao) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 05 Aug 2008 19:20:33 +0200 > From: Jiri Kuthan <jiri(a)iptel.org> > Subject: [Serusers] 2.0 survey > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: <48988BE1.9030503(a)iptel.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I'm just wondering if folks using the 2.0 have found so far any issue > which has not been tracked -- I've been off-line for vacation, IETF and > a couple of other trips and began to be uncertain if I might have missed > something. Any concerns/issues/comments regarding 2.0 are most welcome, > publicly or privately. > > Thanks a lot > > -jiri > > > ------------------------------ > > Message: 2 > Date: Wed, 6 Aug 2008 01:11:34 +0200 > From: "Tomasz Zieleniewski" <tzieleniewski(a)gmail.com> > Subject: [Serusers] ser 2.1 - compilation error - conflicting types > To: serusers(a)lists.iptel.org, serusers(a)iptel.org > Message-ID: > <5fd52d7a0808051611g2017ff27uef0685ef960450d(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-2" > > Hi, > > When I try to compile ser from sources I get the following conflicting > types > errors: > gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops > -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' > -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' > -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.1"' -D__CPU_x86_64 -D__OS_linux > -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' > -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM > -DSHM_MMAP > -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE > -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST > -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK > -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM > -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD > -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM > -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT > -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lock_ops.c -o > lock_ops.o > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:46: error: conflicting types for 'loff_t' > /usr/include/linux/types.h:30: error: previous declaration of 'loff_t' was > here > /usr/include/sys/types.h:62: error: conflicting types for 'dev_t' > /usr/include/linux/types.h:13: error: previous declaration of 'dev_t' was > here > In file included from /usr/include/sys/types.h:133, > from ut.h:53, > from lock_ops.c:28: > /usr/include/time.h:105: error: conflicting types for 'timer_t' > /usr/include/linux/types.h:22: error: previous declaration of 'timer_t' was > here > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:198: error: conflicting types for 'int64_t' > /usr/include/linux/types.h:98: error: previous declaration of 'int64_t' was > here > /usr/include/sys/types.h:204: error: conflicting types for 'u_int64_t' > /usr/include/linux/types.h:97: error: previous declaration of 'u_int64_t' > was here > In file included from /usr/include/sys/types.h:220, > from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/select.h:78: error: conflicting types for 'fd_set' > /usr/include/linux/types.h:12: error: previous declaration of 'fd_set' was > here > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:235: error: conflicting types for 'blkcnt_t' > /usr/include/linux/types.h:124: error: previous declaration of 'blkcnt_t' > was here > make: *** [lock_ops.o] B??d 1 > > Is there any fix for that? > > Regards > Tomasz >

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[Kamailio-Users] Question about registrar lookup() and branching.
by Alex Balashov 06 Aug '08

06 Aug '08

Greetings, I've got a small problem with how registration lookups work. I have several concurrent registrations from one PBX that have multiple contact URIs that they declare that aren't the same; they contain the DIDs to route a call to. So, for instance, in Asterisk: register => user:pass@registrar/DID1 register => user:pass@registrar/DID2 Of course, lookup() doesn't care what those contacts actually are. If it sees multiple contacts for one username, t_relay() will simply create branches for all of them and and ring them all. I can't actually have that happen. I'm looking for suggestions on how to handle this most elegantly. The two options I've got in mind are: 1. Stop using lookup() and do my own URI rewrites and branch forking (if the contact URIs are identical) with attention to the contact. 2. Use BRANCH-ROUTEs to arrest the call leg if it is going to an RURI user part that does not match the user part of the contact URI, and... drop() the branches? I also need to build in some other logic, such that for example if a user has only one contact registered and it does not match the DID, or none of the contacts match the DID, proceed as normal anyway. Is it possible there is a better way? Recommendations welcome. Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Authentication problems.
by Buzukelis 06 Aug '08

06 Aug '08

*Hi all, ** I'm trying to register few ata's to openser.* *i puted two registrations to subscribers table.* *but registaration allwais unauthorized.* ** ** *ngrep output: * *interface: eth0 (192.168.27.0/255.255.255.0) filter: (ip) and ( port 5060 )* *U 192.168.27.117:10186 -> 192.168.27.201:5060 REGISTER sip:192.168.27.201 SIP/2.0. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK645148049;rport. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2000 REGISTER. Contact: *. Max-Forwards: 70. User-Agent: Grandstream HT-502 V1.0B 1.0.1.8. Expires: 0. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Length: 0. .* * U 192.168.27.201:5060 -> 192.168.27.117:10186 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK645148049;rport=10186. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b963. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2000 REGISTER. WWW-Authenticate: Digest realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.27.117:10186 -> 192.168.27.201:5060 REGISTER sip:192.168.27.201 SIP/2.0. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK142179605;rport. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2001 REGISTER. Contact: *. Authorization: Digest username="1000", realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7", uri="sip:19$ Max-Forwards: 70. User-Agent: Grandstream HT-502 V1.0B 1.0.1.8. Expires: 0. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Length: 0. . U 192.168.27.201:5060 -> 192.168.27.117:10186 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK142179605;rport=10186. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.3c26. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2001 REGISTER. WWW-Authenticate: Digest realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. openser.cfg :# # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # ------------------ module loading ---------------------------------- #set module path mpath="//lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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_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) # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest. if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; }; consume_credentials(); lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } can not understand why this configuration is not working to me??? any help will be apreciated. *

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[Kamailio-Users] Change the INVITE timer after append_branch()
by Jens Carl 06 Aug '08

06 Aug '08

Hello List, I have call forwarding on no answer and the possibility to set the time out for the call. I do this by setting the value of "fr_inv_timer" with "avp_fr_timer_avp". Then when the call is route to user's phone and he has call forwarding on no answer activate I overwrite the default INVITE timer with his value. If the call is answered with the error message "Request Timeout (408)" I replace the RURI with the new number, call append_branch() and relay the message to myself. Now the message goes again to my routing logic and will be relayed. I also set a new timeout for the INVITE message, but still the value of the customer is used. For example if the users has a ring time of 10 seconds the second call also rings only for 10 seconds. How can I set the ringtime after append_branch() to my default value again? Is this possible or should I change me routing logic to handle this kind of feature? When I have to change it what is the best way to do this? Thanks in advance Jens

2 2

[Kamailio-Users] 1.4.0 release plans, project matters
by Henning Westerholt 06 Aug '08

06 Aug '08

Hi all, Bogdan-Andrei Iancu prepared this fork from Kamailio (formerly known as OpenSER) since July. This was done secretly, we were not aware of this. We've all done a great amount of work to resolve the existing trademark issues and create a better organisational backing for the project. Many contributors tried hard to improve the quality of the code base, a big number of bug and documentation fixes were done, doxygen code documentation was greatly extended. Because of the project renaming our release schedule needed some adaptions. We discussed this in the last week on the developer list and found a new release date for Kamailio [1]. Version 1.4.0 will be released on the thursday, the 7. august. Cheers, Henning Westerholt [1] http://lists.kamailio.org/pipermail/devel/2008-July/014926.html

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[Serusers] SER/FIFO not accessible
by cindy.cao 06 Aug '08

06 Aug '08

2008-08-06 cindy.cao >Hello all: > >I am trying to add a user to mysql database with the command: > serctl add 1000 1000 1000(a)ser.com > but I get an error: SER/FIFO not accessible! >and i consulted some documents which said it should be the permission >problem. so i insert a line "fif0_mode=0777" into my "ser.cfg" and try the >command again, but now my system halt! > > Any pointers to where to start debuging this error? > Thanks so much > > Best Regards. >Cindy

1 0

[Kamailio-Users] call for contribution: project logo
by Daniel-Constantin Mierla 05 Aug '08

05 Aug '08

Hello, I already received the first proposal, courtesy of Darren Sessions. You can check at: http://www.kamailio.org/pub/kamailio-logos/ This message is an invitation for all of you to submit proposals for the new project logo. We are aware that the name is not making happy many people out there and we can work to solve it, maybe over the time will not look that bad or just send name suggestions. However, this should not stop you being creative and help the project, in a transitional or definitive state. Looking forward to your contributions. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] 2.0 survey
by Jiri Kuthan 05 Aug '08

05 Aug '08

I'm just wondering if folks using the 2.0 have found so far any issue which has not been tracked -- I've been off-line for vacation, IETF and a couple of other trips and began to be uncertain if I might have missed something. Any concerns/issues/comments regarding 2.0 are most welcome, publicly or privately. Thanks a lot -jiri

1 0

[Kamailio-Users] Openser version to run Mediaproxy 2.0
by Nuno Marques 05 Aug '08

05 Aug '08

Hello, I was reading the README file on the download page of Mediaproxy and found this statement: This version of MediaProxy (2.0 or higher) cannot be used in combination with any version of OpenSER older than 1.4 or any components of MediaProxy older than 2.0. You must completely upgrade any previous installation of OpenSER and MediaProxy to these new versions. What i want to ask you is if this is really true because Openser 1.4 (kamailio) isn't available yet... Can i use Mediaproxy 2.0 with Openser 1.3.2 or i need to wait for the Openser 1.4? Thanks in advance, Nuno Marques

2 1

[Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008
by Daniel-Constantin Mierla 05 Aug '08

05 Aug '08

Hello Everybody, as many expressed their intention for an openser/kamailio event in Bucharest, next week, I am going to organize a dinner in a place with nice Romanian beer, wine and food. Seeing the suggestions for the date, I am proposing August 11, 2008 (Monday) -- let me know if you have opinions. Exact hour and place will be sent as soon as I get an idea about number of participants. So far, expressed the intention to participate: - Henning Westerholt, developer and board member - Elena-Ramona Modroiu, developer and board member - Daniel-Constantin Mierla, developer and board member - Michiel ten Hagen - Sebastian Schuman Please write me if you want to attend, I need to know the number of attendants to make the reservation. The event is in the same fashion as the ones happening in Barcelona, Vienna, Berlin, ..., in the past -- everybody pays for its expenses. If you want to have some preamble (might not be same place, but...), you can check some photos from last similar event in Bucharest after the ROSDEV2008 conference: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/index1.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 05 Aug '08

05 Aug '08

David, After double checking, I do seem to be getting E.164 formatted numbers coming from the Cisco datasource, at least for the numbers we care about. Following is an example, you can see that I am able to successfully match a 1800 # as 1800, test (I just can't rate a price to it). id,StartTime,Username,CallerParty,Destination,DestinationId,DestinationName,Duration,Price,Gateway,KBIn,KBOut,Disconnect,Channel,Direction,CallType 1,2008-08-05 08:55:28,,5635833713,2415453,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/2:D:2,outgoing,Telephony 2,2008-08-05 08:55:28,,5635833713,2415453,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/0:D:1,incoming,Telephony 3,2008-08-05 08:55:11,,6082516000,19206750230,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/1:D:22,outgoing,Telephony 4,2008-08-05 08:55:11,,6082516000,19206750230,,,0,,,0,0,0,Normal release,,incoming,VoIP 5,2008-08-05 08:54:37,,6082888583,18004266918,1800,test,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/1:D:18,outgoing,Telephony On Fri, Aug 1, 2008 at 8:23 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > correct > > On Fri, Aug 1, 2008 at 8:11 PM, Brian Del Shasta <briandelshasta(a)gmail.com > > wrote: > >> Okay, I'm using a Cisco datasource, so I'm assuming that would replace >> OpenSER in the general schema, correct? >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[Kamailio-Users] Problem With SipTrace Module
by Ali Jawad 05 Aug '08

05 Aug '08

Hi I did enable the siptrace module to be able to see SIP packets in TLS mode. Once I did set it up, I could not logon to the server anymore. The log file does not show anything : Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=acc Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=nathelper Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=siptrace Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: connection type is xxxxxxx via TCP/IP Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: protocol version is 10 Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: server version is 5.0.48 Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=xlog Aug 5 10:28:21 [30793] DBG:xlog:child_init: init_child [-4] pid [30793] Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=acc Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=nathelper Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=siptrace And the server is listening correctly. # ----------------- SIP Trace Info modparam("siptrace", "db_url", "mysql://mysql://openser:xxxxxxxxx@xxxxxxxxxxxx/openser") modparam("siptrace", "trace_on", 1) Regards

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[OpenSER-Users] XMPP/SIP-SIMPLE Problem
by Cesar Troncoso 05 Aug '08

05 Aug '08

Hi everyone, I have a problem with xmpp module. Messages going from xmpp to sip-simple work fine. The problem I have is in the opposite direction (from sip-sipmle to xmpp). If I set fork=no or children=1 in my config file then all messages are sent and received in both directions. This is what I see in the logs: xmpp: invalid content-type 0x300ff xmpp: unable to write to command pipe: Bad file descriptor Any Idea how to overcome this situation? Thanks! Cesar

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[Kamailio-Users] Message authorization via XCAP
by Adrian Georgescu 05 Aug '08

05 Aug '08

I put this on OpenXCAP roadmap. Regards, Adrian Yann Derichard schrieb: > Hi, > > I have already put in place presence via XCAP and now I would like to > add instant messaging to my system so I would like to know if it is > possible to use OpenSER to do MESSAGE authorization via XCAP (using > OpenXCAP). Hmm - interesting question. It would be good to have a generic framework to check SIP requests against privacy rules stored in XCAP server. does anyone have more knowledge about this? regards klaus

1 0

Re: [Kamailio-Users] Debugging Openser v1.4 scripts
by Jim Burke 05 Aug '08

05 Aug '08

Hi Daniel, After a lot of trial and error I have worked it out! The following portion of script causes the error in both 1.3 and 1.4, however not in 1.2. Root cause is that I have not added "" in my case statements. I.e. case 404: should be case "404": The var reply_status is set from PV $rs in a on_reply or on_failure route using $var(reply_status) = $rs;. switch($(var(reply_status))) { case 404: #Not Found xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Terminating UA Not Found R4\n"); seturi("sip:rva404@domain"); $(avp(early_code)) = "404"; $(avp(early_reason)) = "Not Found!"; append_hf("P-App-name: early_announcement\r\n"); append_hf("P-App-Param: Final-Reply-Code=$(avp(early_code));Final-Reply-Reason=$(avp(early_reason)); "); break; SNIP default: sl_send_reply("$(var(reply_status))","$(var(reply_reason)))"); xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Default Route 4\n"); exit; break; } Let me know if you consider this a bug and I will open a ticket. Regards, Jim -----Original Message----- From: Jim Burke [mailto:jim@evolutiontel.net] Sent: Saturday, 2 August 2008 11:28 PM To: miconda(a)gmail.com Cc: users(a)lists.kamailio.org Subject: RE: [Kamailio-Users] Debugging Openser v1.4 scripts Hi Daniel, Have been busy, will send yuo the config shortly. For my information how do you plan to find the corrupt line. Is there something in GDB that can help us set a tracepoint and identify it? Regards, Jim -----Original Message----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Wednesday, 30 July 2008 12:03 AM To: Jim Burke Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] Debugging Openser v1.4 scripts Hello, On 07/29/08 15:43, Jim Burke wrote: > Hi All, > > We are working to get our script working on Version 1.4 and are having > problems debugging the script. Previously if I saw the following error in > the logs it would highlight the line and colomb numbers were the error > existed. Can someone help out? > > Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, > line 9459656, column 1-1: syntax error > Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, > line 9459656, column 1-1: bad command!) > Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) > it should still show the line and column where error occurred, but here is something wrong with the line. Can you obfuscate the private details in the config (passwords, ip addresses) and send it to me? Cheers, Daniel > Thanks, > Jim > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] Message authorization via XCAP
by Yann Derichard 05 Aug '08

05 Aug '08

Hi, I have already put in place presence via XCAP and now I would like to add instant messaging to my system so I would like to know if it is possible to use OpenSER to do MESSAGE authorization via XCAP (using OpenXCAP). Regards. -- Yann Derichard

2 1

[Kamailio-Users] Cluecon
by Bogdan-Andrei Iancu 05 Aug '08

05 Aug '08

Hi, For people interested in meeting and chatting around the kamilio - opensips subject, in the next 3 days I will be present at ClueCon, in Chicago. So feel free to drop me an email ;) Best regards, Bogdan

1 0

[Kamailio-Users] I don't know why
by Tseveendorj Ochirlantuu 05 Aug '08

05 Aug '08

Dear developers, I don't know what is the reason they will go to two ways. /---- Kamailio OpenSER \---- openSIPS I hope these two project going to powerful. We need OpenSER related projects. I like them. Best regards, Tseveendorj

1 0

[Kamailio-Users] TLS with openser
by Ali Jawad 04 Aug '08

04 Aug '08

hi all Got some issues with tls my pstn gw does not support tls ..but I am using it between my clients and openser. The traffic is being sent encrypted to the pstn gw. I am using sethostport to forward traffic. any ides ? Thx

3 4

[Serusers] cannot create /tmp/ ser_fifo
by 静于 04 Aug '08

04 Aug '08

Hello all, I am trying to add a user to mysql database with the command: serctl add 1000 1000 1000(a)ser.com but I get a error: cannot create /tmp/ ser_fifo Any pointers to where to start debuging this error? Thanks. Best Regards. Ivy --------------------------------- 雅虎邮箱，您的终生邮箱！

2 1

[Kamailio-Users] INVITE branch handling for multiple contacts.
by Alex Balashov 03 Aug '08

03 Aug '08

Greetings, Is there a place somewhere that expounds upon or otherwise illustrates the inner workings of call forking for multiple concurrent contacts? I can only find a patchwork of hints by perusing the documentation for the 'tm' and 'registrar' modules, and they don't provide me entirely with the understanding that I am looking for. Failing that, I am seeking answers and clarifications to the following general questions. All of them concern handling of an INVITE in a situation where the registrar module allows multiple concurrent contacts, and multiple contacts are in fact registered, and the "append_branches" modparam to "registrar" is set to 1. 1. What is officially supposed to happen if multiple contacts to which the INVITE is relayed all answer the call instantaneously, whether via 183 early media or 200 OK -- but not OOB ringing events? What I get seems a little strange, although perhaps there is an explanation that I am missing. Both far-endpoints send a 200 OK w/SDP and I hear both media streams simultaneously when I call, interleaved by clicks. After a while, the second (most recent) contact's media stream drops off because the UAC decides that the call has timed out because it has not received ACK replies to its 200 OKs; at the same time, OpenSER appears to generate a CANCEL for that second call leg. The CANCEL is replied to with a 200 OK (not a 487 Request Terminated?) although this seems a little bizarre since the dialog's state is already established -- but since the 200 OKs are never replied to, I suppose it is not necessary to receive a BYE in order to terminate that request. Meanwhile, 200 OKs in response to INVITE keep coming from the contact that was slower to pick up and was CANCEL'd (why? this is Asterisk 1.4), but the ACKs from them keep being routed to the first contact (the one that remains), which must be understandably confused as to why they're there although processed as retransmissions. I have a packet capture I'd love to send someone for interpretation, but would rather do it privately off-list. 2. Is there a shared REPLY-ROUTE for replies on each branch? Or is there a way to fragment particular reply routes for each branch? The goal I am ultimately looking to achieve is surely commonplace enough; I would like multiple concurrent registrants to be able to be rung at one request URI. But at present, it seems some very strange things are happening when this occurs that confuse the far-end UACs. That's what governs my intuition that I just need to properly understand how branching is supposed to work. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 6

[Kamailio-Users] multidomain test for domains 192.168.0.3 and 192.168.0.32
by Zen Kato 03 Aug '08

03 Aug '08

I have installed openser-1.3.2-nontls on 192.168.0.3. The phones are 0103(a)192.168.0.3, 0203(a)192.168.0.3 and 0303(a)192.168.0.3. To test multidomain support, I added modparam("auth_db|auth_diameter|group|uri_db|usrloc","use_domain",1) and changed from "uri==myself" to "is_uri_host_local" in openser.cfg. I added two phones 9101 and 9102 with domain 192.168.0.32. These two phones are registered on mysql db as follows; # openserctl add 9101(a)192.168.0.32 password 9101@localhost # openserctl add 9102(a)192.168.0.32 password 9102@localhost I also set up 192.168.0.32 for eth0:0. eth0 is 192.168.0.3. Then, I check: [root@p4c8 multidomain]# openserctl domain show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd domain_dump 192.168.0.3 192.168.0.32 FIFO command was: :domain_dump:openser_receiver_6329 [root@p4c8 multidomain]# openserctl domain showdb database engine 'MYSQL' loaded Control engine 'FIFO' loaded +----+--------------+---------------------+---------+ | id | domain | last_modified | version | +----+--------------+---------------------+---------+ | 2 | 192.168.0.3 | 2008-04-09 16:13:12 | 0 | | 4 | 192.168.0.32 | 2008-01-01 00:00:00 | NULL | +----+--------------+---------------------+---------+ [root@p4c8 multidomain]# openserctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=3 max_slot=1 AOR:: 0203(a)192.168.0.3 Contact:: sip:0203@192.168.0.102 Q= Expires:: 295 Callid:: 91741a3a146e34bb(a)192.168.0.102 Cseq:: 101 User-agent:: Grandstream BT110 1.0.8.23 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 AOR:: 0303(a)192.168.0.3 Contact:: sip:0303@192.168.0.103 Q= Expires:: 294 Callid:: 4c0a9450bb744c33(a)192.168.0.103 Cseq:: 101 User-agent:: Grandstream BT100 1.0.6.8 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 AOR:: 0103(a)192.168.0.3 Contact:: sip:0103@192.168.0.101 Q= Expires:: 295 Callid:: 60507d20c32ac791(a)192.168.0.101 Cseq:: 101 User-agent:: Grandstream BT110 1.0.8.23 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 FIFO command was: :ul_dump:openser_receiver_6424 openserctl ul show does not show 9101(a)192.168.0.32 and 9102(a)192.168.0.32 !? [root@p4c8 multidomain]# openserctl ul show 9101(a)192.168.0.32 database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_show_contact location 9101(a)192.168.0.32 404 AOR not found FIFO command was: :ul_show_contact:openser_receiver_6394 location 9101(a)192.168.0.32 What else shoud I set to test multidomain in this situation? Regards, Zen

1 1

[Kamailio-Users] error adding users using openserctl
by muhammad akl 03 Aug '08

03 Aug '08

hey all , i have successfuly built openser-1.2.2-tls from source and till now all things seems to be fine faced a problem with openser_mysql.sh and i have solved it but when i tried to add users to the DB using openserctl i got this message : database engine 'MYSQL' loaded Control engine 'FIFO' loaded /sbin/openserctl: no: not found /sbin/openserctl: no: not found usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] is_user: user counter= Error: user 'zzzz' already exist however i havn't added any users before so any help are more welcomed BR muhammad

1 1

[Serusers] LCR question
by Deepak 03 Aug '08

03 Aug '08

Hi can someone please explain me how LCR works. Specifically, scenario: prefix 1514 handled by gw_1 and gw_2 all other prefixes handled by gw_3 1) Call comes w/ RURI = 1514111111. 2) load_gws is called. In the database, prefix 1514 is supposed to be handled by gw_1 and gw_2; 3) next_gw() is called. RURI is now 1541111111@gw_1 4) call routed to gw_1; gw_1 rejects call 5) we go to failure route and call load_gws() again -- [QUESTION: what happens here?? how does SER know that we need to replace URI w/ 151411111@gw_2 ] 6) next_gws() called. I assume call will go to 1541111111@gw_2 7) gw_2 rejects call. We again go to failure_route. Here again we do a load_gws(). [QUESTION: How does SER know that both gw_1 and gw_2 options have been utilized?? Does load_gws return an error here or does it load gw_3? ] My problem is that in my case, gw_3 is being used after gw_1 and gw_2 have been used. As per my understanding this should NOT happen since prefix 151411111 is not handled by gw_3. Any pointers appreciated.. Running ser 0.9.6

1 0

[Kamailio-Users] Debugging Openser v1.4 scripts
by Jim Burke 02 Aug '08

02 Aug '08

Hi All, We are working to get our script working on Version 1.4 and are having problems debugging the script. Previously if I saw the following error in the logs it would highlight the line and colomb numbers were the error existed. Can someone help out? Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: syntax error Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: bad command!) Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) Thanks, Jim

2 2

[Kamailio-Users] T.38 not relayed by RTPproxy
by Jens Carl 02 Aug '08

02 Aug '08

Hello List, I have problem with T.38 and the RTPproxy by one of mine scenarios. The first one works flawless and looks like that: Linksys ATA -> NAT -> OpenSer/RTPproxy -> T.38-GW (Asterisk) The second one on the other hand don't work how it should and looks like this one: T.38-GW -> SIP-Proxy -> Openser/RTPproxy -> NAT -> Linksys ATA. The normal RTP stream is opened and they switch over to T.38, but the RTPproxy isn't sending out the received T.38 packets from the ATA. If the OpenSER is getting a re-INVITE I call the function force_rtp_proxy(). I already tried the function with 'l' parameter but then none of the scenarios are working. Why is the RTPproxy not relaying like in the 1st scenario? Which flag should I use to get the things working? Why is RTPproxy using the same ports like it used for the normal RTP traffic? I thought when I call force_rtp_proxy() a 2nd time I create a new session with different ports? Can I force the RTPproxy to use different ports? If someone needs a PCAP-trace I can provide this. Many thanks Jens

4 7

Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 02 Aug '08

02 Aug '08

I see. So in my case, since I'm using freeradius, I would need to instruct freeradius to send out the call in E.164 format? I'll look into that... On Fri, Aug 1, 2008 at 3:45 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > I mean instruct OPENSER to send out the call like that, and in cdrtool set > the areacode 1305 and description usa-miami. I'll get you the tables, but > I'll need a couple of days as I'm on the road right now. > > d > > > On Fri, Aug 1, 2008 at 2:44 AM, Brian Del Shasta <briandelshasta(a)gmail.com > > wrote: > >> By "sending out the call in E.164 format" do you mean instructing the >> gateway (or something other than CDRTool) to use that format? And are you >> saying to set the destination id to '1305 USA-MIAMI' or to just '1305' and >> the description to USA-MIAMI? I don't suppose by any chance you have access >> to still and can include an example line from your Destinations, Customers, >> Profiles, and Rates tables? I appreciate your help! >> >> >> On Thu, Jul 31, 2008 at 10:04 PM, David Villasmil < >> david.villasmil.work(a)gmail.com> wrote: >> >>> The only way I got it to work was by sending out the call in E.164 >>> format, i.e. +1035 and setting that same areacode in cdrtool like: 1305 >>> USA-MIAMI... remember pricing is done like 10000 for 1 dolar/euro >>> >>> david >>> >>> On Thu, Jul 31, 2008 at 10:55 PM, Brian Del Shasta < >>> briandelshasta(a)gmail.com> wrote: >>> >>>> >>>> I'm sorry, I don't understand. Do you mean within CDRTool or like on >>>> our actual gateway which feeds info to the freeradius db? >>>> >>>> On Thu, Jul 31, 2008 at 4:41 PM, David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> You MUST set the dialed numbers as E.164 ( I think that's the number ), >>>>> like +1305 for Miami, US. or else you'll never get it working... >>>>> >>>>> d >>>>> >>>>> >>>>> On Thu, Jul 31, 2008 at 11:47 AM, Brian Del Shasta < >>>>> briandelshasta(a)gmail.com> wrote: >>>>> >>>>>> Ok, I changed the time zone to America/Chicago. If anyone else has >>>>>> any luck setting up ratings within CDRTool, I would love to speak with you, >>>>>> thanks... >>>>>> >>>>>> >>>>>> On Thu, Jul 31, 2008 at 10:18 AM, Adrian Georgescu < >>>>>> ag(a)ag-projects.com> wrote: >>>>>> >>>>>>> Central Time is not s valid timezone >>>>>>> Further than this your rates look ok at first glance, if it does not >>>>>>> work maybe other can help with their experiences. >>>>>>> Adrian >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users(a)lists.kamailio.org >>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)lists.kamailio.org >>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)lists.kamailio.org >>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

2 2

[Kamailio-Users] enum module in openser
by Ali Soltani 01 Aug '08

01 Aug '08

Hi, I am using Enum Module in opencer. Here is the piece of the code I am using in openser.cfg: route { ############################################################## if (method=="INVITE") { xlog ("This is the ReqUri as it arrives to openser: $ru"); # before adding + subst_uri ('/^sip:([0-9]+)/sip:+\1/'); xlog ("This is the ReqUri as it arrives to openser: $ru"); # after adding + enum_query("labenum.e164.arpa.com","10.0.0.1"); xlog("This is the return code from the enum query: $rc"); # returns -1 xlog("This is your new ReqUri: $ru"); # no URI change ..... } } Even though the response from Enum query is being sent to openser, enum_query returns -1. enum_query also does not change the current URI. here is the format of enum entry in the dns server: # dig 0.0.0.1.4.8.4.3.1.8.labenum.e164.arpa.com @10.0.0.1 +short 100 10 "U" "E2U+sip" "!^.*$!sip:+18134841000@bts.voice.net!" . Any suggestion? Thanks,

2 1

[Kamailio-Users] About TLS
by Ali Jawad 01 Aug '08

01 Aug '08

Hi All I have Openser running on both tls and udp. On upd it works fine. However when I do enable TLS it connects and I can make call however when I close the call from the softphone the cell phone keeps ringing. If I do close the call from the cell phone the softphone redials the number and calls the cellphone again. I know you guys might want to see a trace for the call however on the server side I did execute : ngrep port 443 and host 87.236.144.9 .......f.\..o..g.g`....W.^%$jp3.TI...zi...f7..8........j.....KV2^>.....M JG.D........S..M..bF...2..qn..b..7g.).J:....!...&.;P?&<..X....v.z.:._.[. v...JMt.4....A%Q...K..W...aD?R ..=."=v.R...m.5G...(V..k6..si.....'.. 9<X.+.....y..P.#..[..GI(<..$<.......4."#.L[....{..Z.......$h..t3zy<.=... . pJ.../......^y.lO.i &\.Q..aA......j('.j.v...{6..m.S.........J..;.6....!......?..W.o...{*.... |..c7...OaTR...O.]R....=t4...m|d..]....7.6w&\.S..62..N.S.T.6m.2.]..x|9]. 2..]r...C..'.zW.,.....z#..l.S.9.I.JD-1.j .q..%.......t..."C..L3...Y!%3$.....)..y.. As you can see the output is encrypted. Shouldn't the output be clear text ? Thanks

3 2

Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 01 Aug '08

01 Aug '08

By "sending out the call in E.164 format" do you mean instructing the gateway (or something other than CDRTool) to use that format? And are you saying to set the destination id to '1305 USA-MIAMI' or to just '1305' and the description to USA-MIAMI? I don't suppose by any chance you have access to still and can include an example line from your Destinations, Customers, Profiles, and Rates tables? I appreciate your help! On Thu, Jul 31, 2008 at 10:04 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > The only way I got it to work was by sending out the call in E.164 format, > i.e. +1035 and setting that same areacode in cdrtool like: 1305 USA-MIAMI... > remember pricing is done like 10000 for 1 dolar/euro > > david > > On Thu, Jul 31, 2008 at 10:55 PM, Brian Del Shasta < > briandelshasta(a)gmail.com> wrote: > >> >> I'm sorry, I don't understand. Do you mean within CDRTool or like on our >> actual gateway which feeds info to the freeradius db? >> >> On Thu, Jul 31, 2008 at 4:41 PM, David Villasmil < >> david.villasmil.work(a)gmail.com> wrote: >> >>> You MUST set the dialed numbers as E.164 ( I think that's the number ), >>> like +1305 for Miami, US. or else you'll never get it working... >>> >>> d >>> >>> >>> On Thu, Jul 31, 2008 at 11:47 AM, Brian Del Shasta < >>> briandelshasta(a)gmail.com> wrote: >>> >>>> Ok, I changed the time zone to America/Chicago. If anyone else has any >>>> luck setting up ratings within CDRTool, I would love to speak with you, >>>> thanks... >>>> >>>> >>>> On Thu, Jul 31, 2008 at 10:18 AM, Adrian Georgescu <ag(a)ag-projects.com>wrote: >>>> >>>>> Central Time is not s valid timezone >>>>> Further than this your rates look ok at first glance, if it does not >>>>> work maybe other can help with their experiences. >>>>> Adrian >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)lists.kamailio.org >>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[OpenSER-Users] Need help on Openser + freeradius
by ha do 01 Aug '08

01 Aug '08

Hi i am trying to config Openser to talk to Freeradius i install openser on first PC : 192.168.1.4 i install freeradius on second PC : 192.168.1.5 on the Openser, i change: edit /etc/radiusclient-ng/radiusclients.conf : authserver & accserver 192.168.1.5 edit etc/radiusclient-ng/servers : 192.168.1.5 openser on the FreeRadius, i change: edit clients.conf : listen { 192.168.1.4 ....} but when i make call Openser doesnt talk to FreeRadius does any ideal on it my openser config: modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("auth_db|permissions|uri_db|usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("nathelper","rtpproxy_disable", 1) modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") oute[3] { ## Requests handler if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); # Radius Extra $avp(s:sip_proxy_ip)="192.168.1.5"; }; Thanks

3 2

[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 01 Aug '08

01 Aug '08

Central Time is not s valid timezone Further than this your rates look ok at first glance, if it does not work maybe other can help with their experiences. Adrian

3 3

[Kamailio-Users] Patch for mi_xmlrpc module Makefile
by Sergio Gutierrez 31 Jul '08

31 Jul '08

Hello everybody. A patch has been published on tracker, which fixes a problem related to not define a macro which makes fail the module compilation in Solaris. It has been registered with Request ID: 2034302* * Hope it be useful. Best regards. Sergio Gutiérrez.* *

1 0

[Kamailio-Users] Kamailio Logo
by Darren Sessions 31 Jul '08

31 Jul '08

Off technical topic, I know, but has the Kamailio logo already been designed? - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com http://www.linkedin.com/in/dsessions _____________________________

2 1

[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

You need to add at list one customer entry, associated profiles <----

2 1

[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

I am afraid that rating all calls the same regardless of destination will not work unless you define some destination to match all your traffic. The rating engine must find a destination prefix from which it derives the price. Read the RATING.txt to understand the logic behind it. Adrian >> I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination...

2 1

Re: [Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

Yes You need to add at list one customer entry, associated profiles and finally the rates linked with the profile which contain the actual prices for each destination. You have example rating files that you can modify. Adrian On Jul 31, 2008, at 3:28 PM, Brian Del Shasta wrote: > That's not a problem. I am currently able to successfully match > destinations by looking for anything that starts with a '1'. I have > a destination setup called "Billable" which applies to any > destination beginning with 1. I also have destinations setup for > 1800, 1877, etc to filter out the 800 numbers. > > What I cannot figure out is how to link a rate to a destination. Do > I have to setup a profile and customer as well? Or should I be able > to price calls using only destination and rates? > > On Thu, Jul 31, 2008 at 8:42 AM, Adrian Georgescu <ag@ag- > projects.com> wrote: > I am afraid that rating all calls the same regardless of destination > will not work unless you define some destination to match all your > traffic. > The rating engine must find a destination prefix from which it derives > the price. Read the RATING.txt to understand the logic behind it. > Adrian > >> > I'm attempting to understand how call rating works within CDRTool. > I am > able to link destinations to the calls, but unable to generate > prices. I'm > using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What > I'm > trying to setup seems simple: I have one set rate of $.08/min for any > call > beginning with a 1 (I'll exclude 800 numbers later). This rate should > be > applied globally with no differentiation between caller party or > destination... > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

1 0

Re: [Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

Hi Neill, The outgoing audio works fine. The audio that is not passed is from PSTN -> Phone. The Snom does have symmetrical RTP set. I did have this working about a week ago then it stopped working! How does asterisk deal with NAT, as if I connect the phone directly to Asterisk everything works perfectly! Both Asterisk & Kamailio are on publically accessible IP addresses. Thank you for your help so far. Ross From: neill.wilkinson(a)btinternet.comTo: users(a)lists.kamailio.orgDate: Thu, 31 Jul 2008 13:32:05 +0100Subject: Re: [Kamailio-Users] NAT Problems Which direction is the one way audio in – from the Phone to the PSTN or Vice-Versa – I assume audio outbound is OK and it’s the audio returning to the phone from the PSTN you’re missing? Do you have symmetrical RTP set in the Line settings (under NAT) on the SNOM? Is Kamailio inside or outside the NAT? Neill...;o) Neill WilkinsonPrincipal Consultant Aeonvista Ltd - opening up new ideas From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ross BeerSent: 31 July 2008 12:50To: users(a)lists.kamailio.orgSubject: [Kamailio-Users] NAT Problems I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= if (!mf_process_maxfwd_header("10")) {sl_send_reply("483","Too Many Hops");exit;};if (msg:len >= 2048 ) {sl_send_reply("513", "Message too big");exit;};# NAT detectionforce_rport();if(nat_uac_test("3")){fix_nated_contact();}if (!method=="REGISTER"){record_route();}if (loose_route()) {append_hf("P-hint: rr-enforced\r\n"); route(1);};if (!uri==myself) {append_hf("P-hint: outbound\r\n"); route(1);};if (uri==myself) {if (method=="REGISTER") {if (!www_authorize("", "subscriber")) {www_challenge("", "0");exit;};if (isflagset(5)) {# set branch flag -- when someone will call this user# the INVITE will have branch flag 6 set after lookup("location")setbflag(6); };fix_nated_contact();fix_nated_register();consume_credentials();save("location");exit;};if (!lookup("location")) {sl_send_reply("404", "Not Found");exit;};append_hf("P-hint: usrloc applied\r\n"); };if(method=="MESSAGE"){if (!lookup("location")){sl_send_reply("404", "User Offline");exit;}route(4);}if (method=="REGISTER"){save("location");} route(1);} route[1] {if (subst_uri('/(sip:.*);nat=yes/\1/')){setbflag(6);};if (isflagset(5)||isbflagset(6)) {route(3);}t_on_reply("1");if (!t_relay()) {sl_reply_error();};exit;} route[3] {if (is_method("BYE")) {unforce_rtp_proxy();} else if (is_method("INVITE")){fix_nated_sdp("3"); };if (isflagset(5)){search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');}t_on_reply("1");}route[4]{if (!t_relay()) {sl_reply_error();};exit;} failure_route[2] {if (isbflagset(6) || isflagset(5)) {unforce_rtp_proxy();}}onreply_route[1] {fix_nated_contact();if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) {fix_nated_sdp("2");force_rport(); }search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');} ==================== SIP PACKETS ===========================INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asteriskMax-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxxs=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY ////////////////////SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID ***CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 Find out how to make Messenger your very own TV! Try it Now! _________________________________________________________________ Invite your Facebook friends to chat on Messenger http://clk.atdmt.com/UKM/go/101719649/direct/01/

2 1

[Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection force_rport(); if(nat_uac_test("3")) { fix_nated_contact(); } if (!method=="REGISTER") { record_route(); } if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # the INVITE will have branch flag 6 set after lookup("location") setbflag(6); }; fix_nated_contact(); fix_nated_register(); consume_credentials(); save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; if(method=="MESSAGE") { if (!lookup("location")) { sl_send_reply("404", "User Offline"); exit; } route(4); } if (method=="REGISTER") { save("location"); } route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { fix_nated_sdp("3"); }; if (isflagset(5)) { search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("1"); } route[4] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[2] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { fix_nated_contact(); if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) { fix_nated_sdp("2"); force_rport(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } ==================== SIP PACKETS =========================== INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asterisk Max-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxx s=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY //////////////////// SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID *** CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508 v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP *** t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 _________________________________________________________________ Find the best and worst places on the planet http://clk.atdmt.com/UKM/go/101719807/direct/01/

2 1

[Kamailio-Users] Kamailio and Snom phones
by Jason Penton 31 Jul '08

31 Jul '08

Hi * not sure if someone has encountered this before but I have noticed a problem with Snom phones registering with Kamailio. It seems that the Snom phones register with a contact port of 2049. However, when Kamailio does the usrloc lookup and the corresponding t_relay, a packet dump shows me that Kamailio is trying to send the invite to port 5060. Can anyone shed some light? Kind regards Jason

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[Kamailio-Users] Error in database module
by Schumann Sebastian 31 Jul '08

31 Jul '08

Dear all I use OpenSER with the perl extension to publish presence states for non-SIMPLE/PUBLISH capable devices using an outbound proxy. My idea is to send all SIP messages to an outbound-proxy, that forwards them to the respective registrar/SIP proxy. The outbound proxy uses SIP messages (INVITE, BYE, CANCEL...) as triggers to call a perl script that publishes the presence state to a presence agent using PUA_MI. Currently, registrar, presence agent and the publishing functionality are hosted on one machine. But I think that should not have any impact on the whole idea... All what I wanted to do works fine so far, except that after some time, I get MySQL errors. The Perl script does not use the database, only from OpenSER script, actions are called. I posted below some of the error messages I receive and after, the part of my config performing database queries. Does anyone see a obvious mistake in there? I tried OpenSER lastest branch and current trunk, both produce the error after some time. Thanks for your help! Sebastian Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:pua:db_update: while querying db table pua Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:xcap_client:query_xcap_update: in sql query Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_delete: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:update_db_subs: deleting expired information from database Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:msg_watchers_clean: querying database for expired messages Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:msg_presentity_clean: querying database for expired messages Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:pua:db_update: while querying db table pua Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:xcap_client:query_xcap_update: in sql query Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_delete: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:update_db_subs: deleting expired information from database Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:msg_watchers_clean: querying database for expired messages Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:msg_presentity_clean: querying database for expired messages 203 if (uri==myself) { 204 if (method=="REGISTER") { 205 # Uncomment this if you want to use digest authentication 206 if (!www_authorize("IP", "subscriber")) { 207 www_challenge("IP", "0"); 208 exit; 209 }; 210 211 # query if dialog exists for registering URI 212 avp_delete("*"); 213 if (avp_db_query("SELECT from_uri, 214 to_uri 215 FROM dialog 216 WHERE from_uri='$tu' or to_uri='$tu' or from_uri='$tu:5060' or to_uri='$tu:5060'")) { 217 #xlog("L_INFO", ">>> Dialog active: caller=$avp(i:1) callee=$avp(i:2)\n"); 218 } else { 219 #xlog("L_INFO", ">>> No Dialog active: avp_db_query() returned no records\n"); 220 }; 221 xlog("L_INFO", ">>>>>>>>> REGISTER received ($tu)\n"); 222 if(!is_avp_set("$avp(i:1)")) { # no call active for registering to_uri user 223 xlog("L_INFO", ">>>>>>>>> PUBLISHED\n"); 224 perl_exec("publish"); 225 } else { 226 xlog("L_INFO", ">>>>>>>>> NOT PUBLISHED, call active\n"); 227 } 228 save("location"); 229 exit; 230 } 231 232 if(is_method("SUBSCRIBE|PUBLISH")) { 233 if(is_method("PUBLISH")) { 234 if($si == "IP") { 235 if (!t_newtran()) { 236 sl_reply_error(); 237 exit; 238 }; 239 handle_publish(); 240 t_release(); 241 } else { 242 sl_send_reply("488","Not Acceptable Here"); 243 } 244 } else if(is_method("SUBSCRIBE")) { 245 if (!proxy_authorize("", "subscriber")) { 246 proxy_challenge("", "1"); 247 exit; 248 }; 249 handle_subscribe(); 250 t_release(); 251 }; 252 exit; 253 }; 252 exit; 253 }; 254 255 lookup("aliases"); 256 257 # native SIP destinations are handled using our USRLOC DB 258 if (!lookup("location")) { 259 sl_send_reply("404", "Not Found"); 260 exit; 261 } else { 262 append_hf("P-hint: usrloc applied\r\n"); 263 if(method=="INVITE") { 264 # Store Dialog in db 265 setflag(5); 266 # Session timer 267 setflag(6); 268 if (sstCheckMin("1")) { 269 xlog("L_ERR", "422 Session Timer Too Small reply sent.\n"); 270 exit; 271 } 272 xlog("L_INFO", ">>>>>>>>> INVITE received ($fu busy)\n"); 273 perl_exec("publish"); 274 } 275 }; 276 };

3 5

[Kamailio-Users] FYI: A1 InnovationDays
by Daniel-Constantin Mierla 31 Jul '08

31 Jul '08

Maybe some developer has a brilliant idea involving openser and looks for some support. Also, maybe you will find there ideas that will help you in the future, so you can comment and vote for it. Cheers, Daniel :: Final Call for the International Developer Challenge A1 InnovationDays :: Since almost 4 weeks, the international developer challenge A1 InnovationDays attracted more than 30 ideas from creative minds. The challenge is open to anyone who addresses the grand theme "Open Communication by Open Standards" and can come up with ideas for cool services built on IP-based communication (SIP & IMS). Ideas can be posted on the platform www.a1innovations.at/en/static/a1innovationdays until August 15, 2008. All entries are open for voting and discussing by all users. On August 15, the 10 highest ranked ideas will be short-listed and judged by a panel of recognized experts as well as representatives from the sponsors. End of August, the best teams behind the ideas will be invited to a prototyping event near Vienna in order to realize their ideas and demonstrate their potential. The winning team can choose among several awards, such as a week of powder snow skiing in Austria. After that, the sponsor (mobikom Austria) is keen to bring your idea to the market. All submissions must be based on Open Source Software. : More Information : For all details about this challenge, go to the FAQs: http://www.a1innovations.at/en/static/faq For registration, go directly to: http://www.a1innovations.at/registrierung : Important dates ahead : August 1, 2008: Two prizes for the best interim ideas will be awarded August 15, 2008: End of idea entries (CET 24:00) August 18, 2008: Announcement of the five selected teams for the prototyping event August 29 - September 1, 2008: Prototyping event at the monastery Und (near Vienna: http://www.und.at/) Other notes about the event: http://blog.tmcnet.com/blog/tom-keating/voip/a1-innovationdays-contest-look… -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Detecting Two Phones Behind Same NAT
by Ross Beer 31 Jul '08

31 Jul '08

Hi, Is there a way to detect two phones behind the same NAT device? Thanks in advance, Ross _________________________________________________________________ The John Lewis Clearance - save up to 50% with FREE delivery http://clk.atdmt.com/UKM/go/101719806/direct/01/

3 3

[OpenSER-Users] Firewall
by CSB 31 Jul '08

31 Jul '08

I have an error which is driving me crazy: Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:core:udp_send: sendto(sock,0x81aaed8,825,0,0xb61216f0,16): Operation not permitted(1) Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:tm:msg_send: udp_send failed This is firewall related because when I turn iptables off the problem goes away. But although all ACCEPT and DENY messages are logged by the firewall, there is no corresponding message logged. When a call is made between two UACs and the far end attempts to answer the call, the call is not answered successfully. The dialogue is shown below. If I turn the firewall off, start a call and then turn the firewall on the call continues successfully. However the next call is not successful. We use Mediaproxy and I suspect that is trying to do some communication that is being blocked by the firewall but that is only a suspicion. The dialogue for this call is shown at the bottom. When comparing these two dialogues with the firewall turned on: - when the callee answers, OpenSER receives a 200 OK but doesn't immediately pass that on to the caller - after a number of 200 messages from the callee, OpenSER sends the 200 to the caller but the Contact is the private IP address whereas when the firewall is turned off it is the public IP address I have also included the iptables config. Note that some lines are commented out due to the testing I'm doing and IP addresses have been changed. Any advice on the changes required would be appreciated. Thanks **************************** SIP dialogue with firewall on U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2222 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D;rport=5060;received=58.28.001.0 01..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35982 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 192.168.1.124:5065ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=566CA8D1-4C8E0458>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK4b4d0d4aF803AD55..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************************** SIP dialogue with Firewall off U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2224 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366;rport=5060;received=58.28.001.00 1..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=3D7BF99A-F3B3ACE5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.2..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************** iptables config $IPTABLES -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A OUTPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT # # Rule 0 (tun0,tun1,tun2) # echo "Rule 0 (tun0,tun1,tun2)" # # # $IPTABLES -N In_RULE_0 $IPTABLES -A INPUT -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A In_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A In_RULE_0 -j ACCEPT $IPTABLES -N Out_RULE_0 $IPTABLES -A OUTPUT -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A Out_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A Out_RULE_0 -j ACCEPT $IPTABLES -A Out_RULE_0 -j ACCEPT # # Rule 1 (lo) # echo "Rule 1 (lo)" # # # $IPTABLES -N In_RULE_1 $IPTABLES -A INPUT -i lo -p icmp -m icmp --icmp-type any -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j In_RULE_1 #$IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_1 $IPTABLES -A In_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A In_RULE_1 -j ACCEPT $IPTABLES -N Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p icmp -m icmp --icmp-type any -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j Out_RULE_1 #$IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_1 $IPTABLES -A Out_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A Out_RULE_1 -j ACCEPT # # Rule 2 (eth0) # echo "Rule 2 (eth0)" # # # $IPTABLES -N Out_RULE_2 $IPTABLES -A OUTPUT -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A FORWARD -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A Out_RULE_2 -j LOG --log-level warning --log-prefix "RULE 2 -- ACCEPT " $IPTABLES -A Out_RULE_2 -j ACCEPT # # Rule 3 (eth0) # echo "Rule 3 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E561900.0 $IPTABLES -A INPUT -i eth0 -d 147.202.001.001 -m state --state NEW -j Cid485B0E561900.0 $IPTABLES -N Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E561900.1 $IPTABLES -N In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 203.89.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 58.28.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 64.38.001.001 -j In_RULE_3 $IPTABLES -A In_RULE_3 -j LOG --log-level warning --log-prefix "RULE 3 -- ACCEPT " $IPTABLES -A In_RULE_3 -j ACCEPT # echo "Rule 4 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E6A1900.0 $IPTABLES -A OUTPUT -o eth0 -s 147.202.001.001 -m state --state NEW -j Cid485B0E6A1900.0 $IPTABLES -N Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E6A1900.1 $IPTABLES -N Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 203.89.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 58.28.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 64.38.001.001 -j Out_RULE_4 $IPTABLES -A Out_RULE_4 -j LOG --log-level warning --log-prefix "RULE 4 -- ACCEPT " $IPTABLES -A Out_RULE_4 -j ACCEPT # # Rule 5 (eth0) # echo "Rule 5 (eth0)" # # # $IPTABLES -N In_RULE_5 $IPTABLES -A INPUT -i eth0 -p tcp -m tcp -m multiport -d 147.202.001.001 --dports 5060,22,443,80,53,25060,8008 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 10000:20000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 35000:36000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport -d 147.202.001.001 --dports 1194,5065,5060,53,10000:20000,35000:36000 -m state --state NEW -j In_RULE_5 $IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_5 $IPTABLES -A In_RULE_5 -j LOG --log-level warning --log-prefix "RULE 5 -- ACCEPT " $IPTABLES -A In_RULE_5 -j ACCEPT # # Rule 6 (eth0) # echo "Rule 6 (eth0)" # # # $IPTABLES -N Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p tcp -m tcp -m multiport -s 147.202.001.001 --dports 22,53,80,443,5060,8008,25060,25 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 10000:20000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 35000:36000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport -s 147.202.001.001 --dports 53,1194,5060,5065,1813,123,10000:20000,35000:36000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A Out_RULE_6 -j LOG --log-level warning --log-prefix "RULE 6 -- ACCEPT " $IPTABLES -A Out_RULE_6 -j ACCEPT # # Rule 7 (global) # echo "Rule 7 (global)" # # # $IPTABLES -N RULE_7 $IPTABLES -A OUTPUT -j RULE_7 $IPTABLES -A INPUT -j RULE_7 $IPTABLES -A FORWARD -j RULE_7 $IPTABLES -A RULE_7 -j LOG --log-level warning --log-prefix "RULE 7 -- DENY " $IPTABLES -A RULE_7 -j DROP # #

2 1

[Kamailio-Users] wildcard in avp_db_load with db_schema
by Andreas Granig 31 Jul '08

31 Jul '08

Hi, For loading all AVPs from "normal" avp tables, I can use something like this: avp_db_load("$fu/domain", "*/domain_preferences"); But when I want to use a db schema to load AVPs belonging for example to a call_id for some sort of dialog tracking, it fails to start with the following error message: "ERROR:avpops:parse_avp_db: inconsistent usage of DB scheme without complet specification of AVP name" My config looks like that: modparam("avpops","db_scheme", "my_schema:table=some_table;uuid_col=call_id;value_col=value") ... avp_db_load("$ci", "*/$my_schema"); If I put something like "$avp(i:10)" instead of the "*", it works, but then I just get this specific AVP instead of all of them. Any idea if this is supposed to work, and if so, how? Thanks, Andreas

2 2

[Kamailio-Users] t_newtran/t_reply vs. sl_send_reply
by mayamatakeshi 31 Jul '08

31 Jul '08

Hello, sorry if this is a stupid question, but in the openser.cfg that comes with the source, I see the following: if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } Could you tell me why we use t_newtran/t_reply for cases -1,-3 and sl_send_reply for case -2. I read docs for modules sl and tm but I still don't get it why we shoud created a transaction in one situation and not in the other. Regards, takeshi

2 1

[Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 30 Jul '08

30 Jul '08

I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination... Any help is greatly appreciated...

1 0

[Kamailio-Users] pua_bla and presence
by Gustavo Mistrinelli 30 Jul '08

30 Jul '08

Hey, I have the same problem , using the same openser configuration file, only adding pua_bla "server_address" parameter on 1.3.2 bla stop working. I installed 2 openser once with 1.3.0 and the other with openser 1.3.2, then start openser using full debug log level to see what happens with each one. I'm using Polycom 650 hardphones Basic console error log ------------------------------------- ERROR:pua_bla:bla_handle_notify: content length= 0 WARNING:tm:t_unref: script writer didn't release transaction http://pastebin.com/m4a1b7969 (OpenSER 1.3.2 debug logs) Full debug log ----------------------- http://pastebin.com/m177c219a (OpenSER 1.3.0 full log) http://pastebin.com/m36f3ffca (OpenSER 1.3.2 full log) The main difference that I saw comparing both logs are remarked in pastbin, and refer to: "DBG:pua:print_subs: pres_uri" Please see below both configuration parameters Openser Parameters ---------------------------------- # Presence server module - common API loadmodule "presence.so" modparam("presence", "server_address", "sip:192.168.4.10") modparam("presence", "max_expires", 3600) # Presence server module - presence & watcher info and XCAP loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) # Common API for presence user agent client loadmodule "pua.so" # BLA extension for PUA loadmodule "pua_bla.so" modparam("pua_bla", "default_domain", "192.168.4.10") modparam("pua_bla", "header_name", "Sender") modparam("pua_bla", "server_address", "sip:192.168.4.10") (Only for OpenSER 1.3.2 I'm sending configuration of affected routes (Main, Register, Presence) so you can see if there is an error. route { # Sanity Check Section #..... # Loose Route Section if (loose_route()) { xlog("L_INFO", "$mi Loose Routing"); if(!has_totag()) { xlog("L_INFO", "OPENSER_NAME: $mi Initial loose-routing rejected - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); route(ACC_FORBIDDEN_ROUTE); return; } if (is_method("CANCEL") || is_method("ACK")) { xlog("L_INFO", "$mi CANCEL/ACK"); route(BASE_LOCAL_ROUTE); return; } if (is_method("SUBSCRIBE") && $rd == "OPENSER_IP") { # in-dialog subscribe requests route(PRESENCE_ROUTE); return; } if(is_method("BYE")) { route(ACC_BYE_ROUTE); xlog("L_INFO", "(ACC) $mi Setting account_flag"); setflag(ACC_FLAG); route(BASE_LOCAL_ROUTE); return; } }; # Call Type Processing Section if (method=="INVITE") { route(INVITE_ROUTE); return; }; if (method=="REGISTER") { route(REGISTER_ROUTE); return; } if(is_method("PUBLISH|SUBSCRIBE")) { route(PRESENCE_ROUTE); return; } if(is_method("NOTIFY")) { if($hdr(event)=="dialog;sla") { route(PRESENCE_ROUTE); return; } } } route[REGISTER_ROUTE] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html xlog("L_NOTICE", "OPENSER_NAME: $mi Register Route - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if ((!www_authorize("OPENSER_DOMAIN", "device")))) { xlog("L_INFO", "OPENSER_NAME: $mi Register Authentication Failed - \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("OPENSER_DOMAIN", "0"); sl_send_reply("401", "Unauthorized"); exit; } else xlog("L_INFO", "OPENSER_NAME: $mi Register Authentication OK -\ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if ( !(src_ip=="LOCAL_PAIR_IP" && src_port==LOCAL_PAIR_PORT) ) { if($hdr(User-Agent) =~ "PolycomSoundPointIP-SPIP_650-UA") { xlog("L_NOTICE", "$mi Polycom BLA set FLAG \ M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); bla_set_flag(); } if (!save("location")) { xlog("L_NOTICE", "$mi $fu save(location) Failed\n"); sl_reply_error(); return(0); }; } consume_credentials(); return; } route[PRESENCE_ROUTE] { # ------------------------------------------------------------------- # Presence Route # ------------------------------------------------------------------- xlog("L_INFO","$mi: Presence Route"); if (!t_newtran()){ sl_reply_error(); exit; }; if(is_method("PUBLISH")){ if($hdr(Sender)!= NULL) { handle_publish("$hdr(Sender)"); } else { handle_publish(""); } t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } else if(is_method("NOTIFY")) { if(bla_handle_notify()) { t_reply("200", "OK"); t_release(); } exit; } Ok, I hope that provided information help to see if there is a problem on my configuration or a is a bug on OpenSER 1.3.2 Thanks in advance, Cheers, Gustavo Mistrinelli

1 0

[OpenSER-Users] OpenSER as NAT traversal proxy HELP!
by Joris Dobbelsteen 30 Jul '08

30 Jul '08

Dear, I'm really trying to use OpenSER as a NAT traversal SIP proxy, since my home phone keeps breaking voice channels (the box was not intended behind NAT and I'm, of course, using a configuration that no so well supported). What is the idea: SIP transactions should travel this way: ZyXEL UA <-> SIP Proxy <-> NAT Firewall (iptables) <-> {Internet} RTP should travel this way: ZyXEL UA <-> NAT Firewall & RTPProxy <-> {Internet} My current test is using X-Lite with voipbuster, but that doesn't really work. It seems that registers are functioning, at least X-Lite reports itself being registered. Voice calls always end up in timeouts, so something is really going wrong here, it might be authentication problems? An added problem is that I have just sufficient knowledge of SIP to see what it is doing, without really knowing what to expect exactly. Furthermore I have virtually no knowledge of OpenSER. I've quite a hard time even grasping the configuration I typed in. This is not really helpful What I do know: * SIP Proxy traffic is flowing. * SIP INVITES don't work at all. * SIP to RTP is communication, but I don't know if RTP is actually flowing. I stole most of the configuration from the "04 NAT Traversal" slides of the "Italy 2007 Admin course", to which there is link on the documentation site. I adapted it to make it work with the debian supplied OpenSER 1.1. How do I get this all working? What am I getting wrong? I really really appeciate any help I can get to get it working! - Joris Config is this: # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) advertised_address="82.168.191.xx" advertised_port=5060 port=5060 children=4 fifo="/tmp/openser_fifo" # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.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_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) # -- nathelper params --- modparam("nathelper", "rtpproxy_sock", "udp:192.168.10.6:22222") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@82.168.191.xx") # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection route(2); # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sip.familiedobbelsteen.nl", "subscriber")) { www_challenge("sip.familiedobbelsteen.nl", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # INVITE will have branch flag 6 set after loopup("location") setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (subst_uri('/(sip:.*);nat=yes/\1/i')) { setflag(6); }; if (isflagset(5) || isflagset(6)) { route(3); }; if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { force_rport(); if(nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { force_rtp_proxy("", "82.168.191.xx"); t_on_failure("2"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[2] { if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); }; } onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status =~ "(183)|(2[0-9][0-9])") { force_rtp_proxy(); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); }; exit; }

7 17

[OpenSER-Users] reply route failure
by Fero Vlkolinsky 30 Jul '08

30 Jul '08

Hello, I am getting the folowing error on communication with one of our application servers: tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 404 This is in situation where I have two triggerings to the AS (orig and term) and term subscriber is not registered to OpenSER. OpenSER thus returns 404 not found which is then sent back to OpenSER to be forwarded to orig leg of the AS. I had the very same behavior with other AS in our lab and found out that it was because of tag added to To header in AS for the 404 response. So i removed the tag and the call worked fine. However with this AS it doesn't seem to work. Is there a way to hack pick_branch so that I see what are the existing branches and why it does not match any? Or is there any easier way to find out the reason? Thanks a lot, Fero --- reklama ----------------------------------------------------- Hľadáš DOM – BYT – POZEMOK – KANCELÁRIU ? http://www.reality.sk

2 1

[Kamailio-Users] Add a table to openser DB
by Diego Costantini 30 Jul '08

30 Jul '08

Hi, I need to use something like this (openser on a pc with 2 network cards): http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy since I couldn't find any documentation to properly create a "location_external" table, I followed this procedure: http://www.tutorialspoint.com/mysql/mysql-clone-tables.htm Now I have the table I need, but when I start openser I get this error: Jul 30 15:49:32 [8374] ERROR:usrloc:register_udomain: Invalid table version (use openser_mysql.sh reinstall) Jul 30 15:49:32 [8374] ERROR:registrar:domain_fixup: failed to register domain Jul 30 15:49:32 [8374] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 451 Jul 30 15:49:32 [8374] CRITICAL:core:fix_expr: fix_actions error Jul 30 15:49:32 [8374] ERROR:core:main: failed to fix configuration with err code -1 Jul 30 15:49:32 [8374] NOTICE:presence:destroy: destroy module ... How can I make Openser happy to accept my new table? Diego

2 1

[Kamailio-Users] Getting SNMPStats to work
by Alex Massover 30 Jul '08

30 Jul '08

Hello! I'm trying to setup SNMPStats on Debian Sid. I have: dev5:~# cat /etc/openser/openser.cfg | grep snmp loadmodule "snmpstats.so" modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 1234) modparam("snmpstats", "MsgQueueMajorThreshold", 5678) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_minor_threshold", 600) dev5:~# cat /etc/snmp/snmpd.conf | grep agent master agentx agentXSocket tcp:localhost:705 dev5:~# cat /etc/snmp/snmpstats.conf agentXSocket tcp:localhost:705 dev5:~# snmpd -f -D -Dagentx -x tcp:localhost:705 2>&1 | less No log handling enabled - turning on stderr logging registered debug token agentx, 1 netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() agentx_register_app_config_handler: registering .conf token for "agentxsocket" agentx_register_app_config_handler: registering .conf token for "agentxperms" agentx_register_app_config_handler: registering .conf token for "agentxRetries" agentx_register_app_config_handler: registering .conf token for "agentxTimeout" /etc/snmp/snmpd.conf: line 115: Warning: Unknown token: system.sysDescr.0. Turning on AgentX master support. agentx/config: port spec: tcp:localhost:705 agentx/master: initializing... agentx/master: initializing... DONE NET-SNMP version 5.4.1 agentx/master: unexpected callback op 4 agentx/master: handle pdu (req=0x365b5eac,trans=0x0,sess=0x0) agentx/master: open 0x86b37e8 agentx/master: opened 0x86c2028 = 6 with flags = a0 agentx/master: send response, stat 0 (req=0x365b5eac,trans=0x0,sess=0x0) But it doesn't work: dev5:~# snmpget -v2c -c public localhost openserSIPEntityType.0 OPENSER-SIP-COMMON-MIB::openserSIPEntityType.0 = No Such Object available on this agent at this OID Am I missing something? -- Best Regards, Alex Massover

1 0

[Kamailio-Users] OpenSER + Asterisk (Voicemail) Issue....
by Gerard A. Matthew 30 Jul '08

30 Jul '08

I'm having this little issue when implementing the voicemail feature. My openser.cfg looks like this in the failure route: if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemail.mydomain.com:5061"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } On my asterisk end after the time out, i'm viewing the following: SELECT * FROM sipusers WHERE name = 'XXXXXX' i.e XXXXXX = the PSTN number i'm using to call into the IP phone that's connected to OpenSER There seems to be a simple mixup with the number that is sent to asterisk. Obviously there is no user with the PSTN number, however there is one with the called number. Any idea as to what wold be causing this? Have I provided enough information? Regards, Gerard.

2 1

[OpenSER-Users] Polycom Phones produce no wringing when dialing pots lines on the pstn
by Alex G 30 Jul '08

30 Jul '08

I'm sure i'm not the first to experience this. My polycom phones when calling out a pstn gateway recieve far end ringing when dialing cellphones in the US, but when i try to call a pots line i get no ringing, only silence until the end point answers the call. Does anyone have a solution for this? From what i can see in the sip captures, the gateway sends a 180 ringing on the pots line calls, but when i call a cellphone i recieve a 183 Session Progress where the phones do provide the ringing. A polycom to polycom call shows the same 180 ringing, but in this case they do provide ringing. Just to be clear, the ringing i am talking about is the rinigng you here in the earpiece after a dial. Any insight or help would be very appreciated. I have sip captures that i can post, but opted not to incase someone has seen this and knows the answer first. Thanks so much!

3 2

Re: [Serusers] use_dns_cache and call forwarding
by caio 29 Jul '08

29 Jul '08

Hi Miklos, This happens only when I try to do a call to a subcriber having forwarding enabled. Normal calls success ok. I'm not sure about the append_branch() usage and how to work with 'on busy' and 'no answer' forwarding features. Thanks your comments.. Claudio On Thu, Jul 24, 2008 at 4:58 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > Hi, > > On 07/23/2008 05:22 PM, caio wrote: >> >> Hi guys, >> I have a problem with call forwarding (on busy or not answer).., the >> call fall into the failure_route(1) block, and AVP checking and pushto >> does its job. >> >> Here a snipped of code of failure_route(1) (flag 27 means call fwd if >> no answer): >> >> if (isflagset(27) && t_check_status("408")) { # if fwd no >> answer is set and reply msg is 408 >> >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> resetflag(27); >> >> avp_print(); >> >> log(1, "LOG: --> appending new branch..."); >> if (!append_branch()){ >> t_reply("500", "Too many branches?!"); >> drop; >> } >> >> log(1, "LOG: --> Calling route_6..."); >> route(6); >> break; >> >> Route(6) finally calls route(1), where t_relay is called. >> But have a dns failure..and the call never is forwarded. >> >> == ser.log == >> >> DEBUG: mk_proxy: doing DNS lookup... >> get_record: lookup(_sip._udp.wsa.lab, 33) failed_ >> sip_resolvehost: no SRV record found for wsa.lab, trying 'normal' >> lookup..._ >> check_via_address(190.244.33.5, 190.244.33.5, 0)_ >> ERROR: udp_send: sendto(sock,0x2ab32599c720,910,0,0x2ab325999e68,16): >> Operation not permitted(1)_ >> msg_send: ERROR: udp_send failed_ >> ERROR: t_forward_nonack: sending request failed_ >> DEBUG: add_to_tail_of_timer[4]: 0x2ab325999e80_ >> DEBUG: add_to_tail_of_timer[0]: 0x2ab325999ea0_ >> ERROR: w_t_relay (failure mode): forwarding failed_ >> >> A question regarding this trouble.....Can I disable use_dns_cache ? >> SER version is 0.9.7. > > there is no DNS cache in version 0.9.x as I recall, but anyway, your problem > does not seem to be related to DNS, but to the UDP message sending: SER > fails to send out the UDP packet, "Operation not permitted". Is there no > firewall or routing problem at the proxy side? > > Miklos > >> What do you recommend? >> If you need my ser.cfg I can paste it... >> >> Thanks.. >> > -- caio -- caio

2 8

[Kamailio-Users] rtpproxy scalability.
by Alex Balashov 29 Jul '08

29 Jul '08

Does anyone have any reliable data on rtpproxy dimensioning, i.e. just about how many calls per box (and what kind of specs) it can really handle in actually-existing practice, not in theory? The one time I used it for NAT traversal was on a pretty beefy quad-CPU box, and I started seeing clicks and distortions in the calls beyond a load of 90 or so. However, I was never able to decisively ascertain that the cause is not the network. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] new list for announcements
by Henning Westerholt 29 Jul '08

29 Jul '08

The Kamailio project has now a dedicated list for announcements. There will be only important messages posted to this list, like release announcements, security advisories, informations about important changes in the project organisation and community events. The posting frequency will be problably not more then a few per month. All postings will be moderated. The address of the list is: kamailio-announce(a)lists.kamailio.org, to subscribe visit: http://lists.kamailio.org/cgi-bin/mailman/listinfo/kamailio-announce Best regards, Henning

1 0

Re: [Kamailio-Users] Prepaid system using openSER
by Darren Sessions 29 Jul '08

29 Jul '08

I’ve used SER and now OpenSER (uhhm, Kamailio) for the sip proxy and cdr generation (along with media proxy and cdrtool) and Asterisk as the ivr piece for years. Using a proxy like OpenSER/Kamailio to front your IVR servers is your best bet. The combination has worked extremely well for me. - Darren -- www.darrensessions.com On 7/28/08 6:52 PM, "Rana Dhekial" <dhekial(a)msn.com> wrote: > Hi, > > We have a requirement to build a prepaid system using OpenSER. I > took a look at CDRTool which looks quite promising. But I am not > sure how to build the Call control module and provide IVR type > information to the end users. > > If any one has done similar type of job, please respond. > > thanks, > > Rana > > Keep your kids safer online with Windows Live Family Safety. Help > protect your kids. <http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa… > >

1 0

[Kamailio-Users] HELP: Create new table for Nathelper
by Diego Costantini 29 Jul '08

29 Jul '08

Hi, I repropose my unanswered question. I was following this sample configuration for a multihomed server: http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy and it mentions: "For this example to work, you must also follow the nathelper module instructions for creating new location_internal and location_external tables. Be careful to use underscores ("_") and not hyphens ("-") when creating the new tables." But I couldn't find any instruction anywhere. Can someone address me to the correct place please? Thanks, Diego

2 4

[Serusers] What is the behavior during Update of Contact List in ser2?
by Tomasz Zieleniewski 29 Jul '08

29 Jul '08

Hi, I have a question concerning registration. If my UA performs a successful registration and during contact update it sends first REGISTER request with the Authorization header already present with new nonce and response params, will ser respond with 401 or OK. Will functions such us radius_www_authenticate("$fd.digest_realm"); and www_authenticate("$td.digest_realm", "credentials"); return success?? BTW how is the avp fd.digest_realm set? Now when we try to do this ser responds with 401. Is this correct? Kind regards Tomasz

1 0

[Serusers] SER and Radus auth + acc
by Samuel Muller 29 Jul '08

29 Jul '08

Hello all, I've a single question : Does anyone made a SER v2.0 with a working Radius authentication and accounting ? I actually am interest to get some tips, I've many problems about AVPs, DB relations, proxy authentication, ... and i don't really find any full updated docs. The config : SER 2.0, mySQL, Radiator. Many thanks in advance ! Samuel MULLER

1 0

[Serusers] retargeting - rerouting - redirecting
by .:: Francesco la Torre ::. 29 Jul '08

29 Jul '08

Dear all, few days ago I found in a script posted in this mail-list a function that allow to forward a sip message to another host in this way function_name("ip.host.target.message","port"); Is there anyone that know the name of this function ? Thank a lot, Flt

2 3

[Kamailio-Users] Prepaid system using openSER
by Rana Dhekial 29 Jul '08

29 Jul '08

Hi, We have a requirement to build a prepaid system using OpenSER. I took a look at CDRTool which looks quite promising. But I am not sure how to build the Call control module and provide IVR type information to the end users. If any one has done similar type of job, please respond. thanks, Rana _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

2 1

[Kamailio-Users] Pronunciation
by Darren Sessions 29 Jul '08

29 Jul '08

So, what's the phonetic pronunciation of 'kamailio' ? :)

1 0

[Kamailio-Users] Project continuity and near future.
by Adrian Georgescu 28 Jul '08

28 Jul '08

Alex Nothing has changed. The King is dead. Long live the King! http://en.wikipedia.org/wiki/The_King_is_dead._Long_live_the_King%21

1 0

[Kamailio-Users] Project continuity and near future.
by Sergio Gutierrez 28 Jul '08

28 Jul '08

Hello to all the members. As user of the project and responsible of an implementation of it in a production large scale environment. it is a little bit worrying reade terms as "conflicting interests" related to project in the news published today. According to this new change within the project, which I feel go further than renaming it, what could the board members to comment about the RoadMap, philosophy and near future of Kamailio (Formerly known as OpenSER)? One of the biggest warranties that users have felt ourselves is exactly the support and knowledge from the founders and leaders, and it would be very worrying not been able to count with it anymore. Thanks in advance for your attention, and I apologize in advance if this question is out off topic. Best regards. Sergio Gutiérrez.

5 7

[Kamailio-Users] [OpenSER-Users] [MediaProxy] local vs. remote domains?
by Adrian Georgescu 28 Jul '08

28 Jul '08

The readme file that comes with the software explains this. Adrian I've been looking at the mediaproxy (v1) code, and am a little confused why it has the concept of "local" and "remote" domains and why it treats them differently. Can anyone shed light on this? Max.

2 1

[Kamailio-Users] [OpenSER-Users] [MediaProxy] If early media is longer than "idleTimeout" the RTP session is terminated
by Adrian Georgescu 28 Jul '08

28 Jul '08

Hi Iñaki, As far as we are concerned we do not maintain MediaProxy 1.x anymore. So is likely this will not get fixed. All new developments and bug fixing are done against 2.0 Adrian >>>>>>> Hi, I use MediaProxy 1.9.1 with OpenSer. idleTimeout=50 seconds I've received an INVITE and a 183 with early media during 52 seconds and after that the 200 was received. But the RTP session in MediaProxy was terminated after 50 seconds (idleTimeout). Why? Is a still no answered call considered "idle"? This is the syslog:

1 0

Re: [Kamailio-Users] [OpenSER-Users] Firewall
by Henning Westerholt 28 Jul '08

28 Jul '08

On Thursday 24 July 2008, CSB wrote: > I have an error which is driving me crazy: > > Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:core:udp_send: > sendto(sock,0x81aaed8,825,0,0xb61216f0,16): Operation not permitted(1) > > If I turn the firewall off, start a call and then turn the firewall on the > call continues successfully. However the next call is not successful. We > use Mediaproxy and I suspect that is trying to do some communication that > is being blocked by the firewall but that is only a suspicion. The dialogue > for this call is shown at the bottom. Hi CSB, how about debugging your firewall rules step by step and test each rule? This way you should be able to isolate the problematic one. Cheers, Henning

1 0

[OpenSER-Users] OpenSER becomes Kamailio Project
by Adrian Georgescu 28 Jul '08

28 Jul '08

July 28, 2008 OpenSER becomes Kamailio Project OpenSER project and the community around it have grown considerably in the past three years. Because of its extensive development and high adoption rate, conflicting interests related to the OpenSER name and similar trademarks emerged. Consequently, we have been forced to find a new name for the project that stands on itself and represent the project in its future expression. Starting with version 1.4, OpenSER is called Kamailio. Existing OpenSER users can keep on enjoying the benefits of the best open source SIP server available on the market under this new name. All project infrastructure like the website and mailing list will be in the future accessible under the kamailio.org domain, but we will keep the old one functional for a transition period to make the switch as easy as possible for you. Kamailio is a hawaiian word. Kama'ilio means talk, to converse. It was chosen for its special flavour. It is hopefully easy to remember and the meaning fits well with the project purpose. We hope you like it too! For more information visit: http://www.kamailio.org -- http://www.openser.org/mos/view/Management/

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