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37563 discussions

SIP over TLS
by aawaise 05 May '14

05 May '14

Hello, I am running a kamailio server with SIPS over TLS. Now I have to connect an android based SIP client to server. The SIP client has no support for SIPS over TLS. But it can run SIP over TLS. First I want to know the difference between the two ?? Secondly how can I make my server accept SIP connection over TLS ? Thanks. Cheers, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/SIP-over-TLS-tp127469.html Sent from the Users mailing list archive at Nabble.com.

2 3

sip trunk
by Joli Martinez 05 May '14

05 May '14

Hello, I am setting up Kamailio as class4 softswitch for wholesale minutes. I am trying to figure out how to add a SIP trunk to kamailio. thanks,

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Kamailio and mysql connection error: db_mysql [km_my_con.c:109]: driver error:
by toasterisk 05 May '14

05 May '14

hello, all of users: I am using this link for kamalio+asterisk+mysql: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb i do everything based on the link, only different thing is I installed Kamalio+mysql at same server, and asterisk in another server. currently, my asterisk can connect with the remove MYSQL server, but kamailio can not connect with that. when start up the kamailio, i got the error message: ------------------------------------------ 0(3739) DEBUG: <core> [sruid.c:100]: root for sruid is [uloc-5365e604-e9b-] (0 / 18) 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): ctl 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): mi_rpc 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): acc 0(3739) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (1): auth_db 0(3739) DEBUG: <core> [db.c:316]: connection 0xb71aab90 not found in pool 0(3739) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/asterisk 0(3739) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'asterisk'@'localhost' (using password: YES) 0(3739) ERROR: <core> [db.c:320]: could not add connection to the pool 0(3739) ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database 0(3739) ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib/kamailio/modules_k/auth_db.so) 0(3739) ERROR: <core> [main.c:1433]: main_dontfork: init_child failed 4(3744) INFO: <core> [main.c:851]: INFO: signal 15 received 3(3743) INFO: <core> [main.c:851]: INFO: signal 15 received 0(3739) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(3739) ALERT: <core> [main.c:782]: child process 3743 exited normally, status=0 0(3739) ALERT: <core> [main.c:782]: child process 3744 exited normally, status=0 0(3739) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(3739) DEBUG: <core> [db_pool.c:97]: connection still kept in the pool 0(3739) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(3739) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(3739) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(3739) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(3739) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(3739) DEBUG: <core> [mem/shm_mem.c:242]: shm_mem_destroy 0(3739) DEBUG: <core> [mem/shm_mem.c:245]: destroying the shared memory lock --------------------------------------------------------------------------------------------------------------------------- [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = 192.168.1.116 // IP for kamailio and mysql USER = asterisk PASSWORD = 111111 PORT = 3306 DATABASE = asterisk -----------------use isql test--------------- root@kamailio:/etc# isql -v ODBC [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect root@kamailio:/etc# ----------------------------------------------------------- anyone knows the problem? thanks!

3 2

Location table not getting updated
by VOIP Tests 04 May '14

04 May '14

Hello, can someone let me know what could be the reason that the location table in the kamailiio db does not get updated? There are some extensions that I can see in the location table but not others, but I am still able to make extension to extension calls to all extensions. This is a multi domain set up. Thank you, Arun

2 5

xmlops & XPath question
by Alex Balashov 04 May '14

04 May '14

Hello, I don't know a lot about XML or XPath, so I was hoping someone could help me out. I'm trying to parse out 'Event: dialog' state notifications from NOTIFYs using xmlops. They look like this: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="5" state="full" entity="sip:abalashov-roaming@xxx.xxx.xxx.xxx"> <dialog id="0588e460-d293-11e3-8563-538b9e69fee5" direction="initiator"> <state>terminated</state> </dialog> </dialog-info> So, I do: $xml(u=>doc) = $rb; xlog("L_INFO", "State: $xml(u=>xpath:/dialog-info/dialog[1]/state/text())\n"); This produces a blank result. I've tried numerous other variations, such as: xpath:/dialog-info/dialog/state/text() None of them work, either. When I upload the above XML body to various online XPath validation tools and try my expressions, they work fine. Any ideas as to what I'm missing? Do I need to indicate the namespace in the XPath expression somehow? If so, how do I do it? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 5

[sr-user] textops module - search_append_body(re, txt) - append new line
by Cockhootec Yahrabee 03 May '14

03 May '14

Hi, I am trying to append new field in SDP and I need to append new line but I don't know how. I tried to google for solution but I didn't find anything. I want to route my RTP and RTCP packet to port 5060 (this is for development of my school project, I know that in the context of correct SIP this has no sense but this is only experimental). My sample kamailio.cfg code: if (is_method("INVITE")) { # route RTP packets to 5060 if (search_body('m=audio [0-9]+')) { subst_body('/^m=audio [0-9]+/m=audio 5060/g'); } # route RTCP packets to 5060 if (search_body('a=rtcp:[0-9]+ IN IP4 [0-9]+.[0-9]+.[0-9]+.[0-9]+')) { subst_body('/^a=rtcp:[0-9]+ IN IP4 [0-9]+.[0-9]+.[0-9]+.[0-9]+/a=rtcp:5060 IN IP4 192.168.11.129/g'); } else { search_append_body('m=audio [0-9]+ RTP/AVP [0-9]+ [0-9]+ [0-9]+ [0-9]+ [0-9]+ [0-9]+ [0-9]+', ' *{NEW_LINE_NEEDED_HERE}*a=rtcp:5060 IN IP4 192.168.11.129'); } } } Could someone please help me resolve what should I replace the {NEW_LINE_NEEDED_HERE} with to add new line after 'm=audio' field? Also If you have some idea how should I optimize my regexp for finding 'm=audio'/'a=rtcp' fields I would appreciate your help.

1 0

Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
by Alex Villacís Lasso 02 May '14

02 May '14

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to come from localhost, from the point of view of asterisk. Currently I have a model on which internal SIP phones get identified by the authentication username, and then the contact names at From: and To: get massaged to incorporate the SIP domain, in order to emulate multiple-domain support. The 'sip' table in Asterisk defines all such contacts as SIP accounts of the form name_domain.com, and the SIP phones are configured to use 'name' as authentication username for domain 'domain.com'. However, SIP providers that register on the server with authentication names are left with their original names, since in the model, SIP trunks are available to all domains. Now I have to add support for SIP providers which are to be authorized on the basis of IP only. Apparently, the permissions.so (WITH_IPAUTH) is made for just this purpose, so I enabled it. After authentication, I need to route the INVITE to asterisk, and asterisk must somehow match the account for the SIP trunk from the available information on the INVITE request. A typical INVITE for this scenario looks like this, before being processed by kamailio: INVITE sip:6008010@172.28.161.218:5060;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.25.144.58:5060;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1 Call-ID: FBE75B3A@32e3da76 From: <sip:042294440@200.25.144.58:5060;user=phone>;tag=32e3da76+1+544c000c+52be771c To: <sip:6008010@172.28.161.218:5060;user=phone> CSeq: 975469826 INVITE Expires: 180 Organization: SetelGYE Min-SE: 90 Session-Expires: 18000 Supported: replaces, 100rel, timer Contact: <sip:042294440@200.25.144.58:5060;transport=udp;user=phone> Content-Length: 149 Content-Type: application/sdp Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE, INFO, REFER v=0 o=- 0 0 IN IP4 201.217.79.3 s=- c=IN IP4 201.217.79.3 t=0 0 m=audio 5388 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Here, 6008010 is the phone number that was dialed at the provider in order to reach my system, and 042294440 is the provider-supplied Caller-ID, which I want to preserve all the way to Asterisk. In particular, 042294440 appears as the value that ends up as $fU (From: username) while being processed in kamailio. If I pass the SIP packet as-is to asterisk, asterisk first tries to match by the value of $fU, which obviously fails to match the trunk name. It then tries to match by incoming IP, which also fails because asterisk received this packet from 127.0.0.1 . Finally, asterisk sort of matches to the first record in the sip table, which is *not* the SIP account for this trunk, but some other random account. I have a partial solution that uses sqlops to make a query to the sip table, using the $si (source IP) and reads the trunk name in order to replace $fU. This works, as now $fU will have the trunk name and asterisk will now recognize the intended SIP account for the trunk. However, this has the unfortunate side effect of throwing out the Caller-ID information. What is the standard/proper way to deal with this situation? Is there a well-known way to make Asterisk match the trunk name, without overwriting the Caller-ID information? Before you ask, requesting the provider to modify its INVITEs is not an option. I believe there is a standard way to deal with this, since this scenario should also arise with a kamailio that faces the internet, and relays INVITEs (after authentication) to an asterisk in a local network. As far as I can tell, the fact that in my case the 'local network' is localhost should be irrelevant.

2 3

Kamailio v4.0.5 crash - transformations related ?
by Dragos Oancea 02 May '14

02 May '14

Hi We experimented a crash with kamailio 4.0.5 , it looks like a memory corruption. After an analyse of the core file, it appears that it crashed while doing a str2int transformation (trying to convert the value of myvar to int): if (($(dlg_var("myvar"){s.int}) == 0) && some_other_condition ) { do_something(); } gdb output: (gdb) frame 4 #4 0x00000000004bcb77 in rval_get_btype (h=0x7fffd69713d0, msg=0x7ffc8a58a4d0, rv=0x7ffc8a3dfc18, val_cache=0x7fffd69706a0) at rvalue.c:418 418in rvalue.c (gdb) i loc r_avp = 0x7fffd69709b0 tmp_avp_val = {n = -1975661232, s = {s = 0x7ffc8a3dcd50 "\034\001", len = -1973902128}, re = 0x7ffc8a3dcd50} avpv = 0x7fffd6970928 tmp_pval = {rs = {s = 0x7ffc8a5b86e0 "route[MAIN]: call-id=52e8c2b553540db4 from=987654321 to=+1234567890 : ACK ip=10.0.x.x", len = -1975658024}, ri = -694745968, flags = 32767} pv = 0x7fffd69706a8 tmp = RV_NONE ptype = 0x7ffc8a58a4d0 __FUNCTION__ = "rval_get_btype" (gdb) p *pv $8 = {rs = {s = 0x7069736f58652820 <Address 0x7069736f58652820 out of bounds>, len = 775106354}, ri = 0, flags = 4} (gdb) p val_cache->c.pval $9 = {rs = {s = 0x7069736f58652820 <Address 0x7069736f58652820 out of bounds>, len = 775106354}, ri = 0, flags = 4} <- the value of s is invalid, it's a string from a SIP message. Full GDB backtrace and info locals here : kamailio 4.0.5 crash - memory corruption ? - Pastebin.com kamailio 4.0.5 crash - memory corruption ? - Pastebin.co... (gdb) bt full #0 0x00007ffc822851e8 in str2sint (_s=0x7fffd69706a8, _r=0x7fffd69706b8) at ../../ut.h:681 i = 0 sign = 1 View on pastebin.com Preview by Yahoo I still have the core file and I can help with further analysis . Regards, Dragos PS: Kamailio still rocks.

2 1

Frequent Crash in Kamailio 4.1.3 in the file data_lump.c see gdb backtrace in mail
by varun pratapsingh 02 May '14

02 May '14

HI All/ Daniel, According to the recommendation I have migrated my Kamailio 4.1.2 to the Kamailio 4.1.3. But very frequent crashes are coming in Kamailio 4.1.3 which was not coming on the 4.1.2. The gdb backtrace of the crash is below: I have seen many crashes in couple of days but the "data_lump.c: No such file or directory" this line is coming in every crash. I think every time it crashes in the same file and same function somewhere else. GDB Backtrace: Program terminated with signal 11, Segmentation fault. #0 0x080791c5 in count_applied_lumps (ll=0xb74f9898, type=10) at data_lump.c:702 702 data_lump.c: No such file or directory. in data_lump.c Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.132.el6.i686 (gdb) where #0 0x080791c5 in count_applied_lumps (ll=0xb74f9898, type=10) at data_lump.c:702 #1 0x00141392 in add_uac (t=0xb5688bc8, request=0xb7506288, uri=0xb75062ac, next_hop=0xb75062ac, path=0xb7506604, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0xb750660c, ruid=0xb7506618, location_ua=0xb7506620) at t_fwd.c:855 #2 0x001459a1 in t_forward_nonack (t=0xb5688bc8, p_msg=0xb7506288, proxy=0x0, proto=0) at t_fwd.c:1721 #3 0x00139a3a in t_relay_to (p_msg=0xb7506288, proxy=0x0, proto=0, replicate=0) at t_funcs.c:354 #4 0x0015d9d3 in _w_t_relay_to (p_msg=0xb7506288, proxy=0x0, force_proto=0) at tm.c:1515 #5 0x0015e41e in w_t_relay (p_msg=0xb7506288, _foo=0x0, _bar=0x0) at tm.c:1713 #6 0x0805d400 in do_action (h=0xbf849ba0, a=0xb74e5194, msg=0xb7506288) at action.c:1099 #7 0x08064dd9 in run_actions (h=0xbf849ba0, a=0xb74e5194, msg=0xb7506288) at action.c:1599 #8 0x08065429 in run_actions_safe (h=0xbf84ae9c, a=0xb74e5194, msg=0xb7506288) at action.c:1664 #9 0x080ff0c4 in rval_get_int (h=0xbf84ae9c, msg=0xb7506288, i=0xbf849f18, rv=0xb74e53ac, cache=0x0) at rvalue.c:924 #10 0x08101a48 in rval_expr_eval_int (h=0xbf84ae9c, msg=0xb7506288, res=0xbf849f18, rve=0xb74e53a8) at rvalue.c:1918 ---Type <return> to continue, or q <return> to quit--- Regards Varun

2 1

Kamalio
by Akash Ashok 02 May '14

02 May '14

Hi, I have just started using Kamalio and We are using it for a VOIP app that I am trying to build. Here are the things that i have done 1. I have setup kamalio server 2. I could get it running using linphone app 3. I could add users using kamctl command Since I am building a smartphone app I want to register users remotely from the phone. I did check up the XML RPC documentation and it doesn't talk about apis to add users and a lot of other functionalities that are present on kamctl commandline tool. I am wondering if there a webservice API tha ti can user to manage users and locations and accounts on kamalio ? -- Cheers, Akash A ᐧ

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