sr-users May 2014

sr-users@lists.kamailio.org

96 participants
155 discussions

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.

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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.

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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

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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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What does this script do?.
by VOIP Tests 02 May '14

02 May '14

Can someone let me know what this script does? avp_db_load("$ru/domain","i:/domain_preferences"); Thank you, Arun

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Limit the number of requests per second based on the user
by Sebastian Damm 02 May '14

02 May '14

Hi, I'm looking for a solution to limit the number of call initiations or OPTIONS packets or REGISTER packets per user. Neither pike, nor pipelimit or ratelimit give me the possibility to do this. Limiting it based on the source IP is not an option in our case. Is there a module I can use? Or is this generally a bad idea? Best Regards, Sebastian <http://www.sipgate.co.uk>

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Dispatcher ping source ip:port?
by Daniel Tryba 02 May '14

02 May '14

Is it possible to manipulate the pings (OPTIONS in my case) to force a source ip:port combo per destination? dispartcher.list: 1 sip:192.168.0.1:5060 2 sip:192.168.1.1:5070 3 sip:192.168.2.1:5050 The kamailio (4.1) is listening on 192.168.3.1 5060 and 5070. dispatcher 2 cares about the source port, which should be 5070. 1 and 3 don't care. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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Re: [SR-Users] Voice is breaking when using Kamailio 4.1.3
by varun pratapsingh 02 May '14

02 May '14

Hi Daniel, Yes I agree with you that kamailio is not involved in the rtp handling and forwarding but RTP Proxy does this. So I have mentioned that when my call is established then my RTP passes through the RTPProxy. The issue is that RTP is coming proper at UAC Side so no Voice Problem but may be RTP is not coming good at Phone Side that is why voice is bad at Phone SIde. Please see again my network. SIP UAC--------->Kamailio+RTPProxy------>SIP Server------->Phone And Kamailio and RTPProxy are on same machine and load is not high only one call is enough to produce this issue. It seems that this issue is with RTPProxy. Please anyone can provide me some settings or inputs on this issue. I can provide more information if required. Thanks and Regards Varun Date: Fri, 02 May 2014 11:27:09 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Voice is breaking when using Kamailio 4.1.3 Message-ID: <536364ED.4080806(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, kamailio is not involved in the rtp handling/forwarding. So either is a bad connection somewhere in the path or an issue in the client sides. Can you try with different phones? Cheers, Daniel On 02/05/14 07:49, varun pratapsingh wrote: > HI All/ Daniel, > We are using the Kamailio+ RTPProxy as a SIP and RTP Proxy to call the > Phones. Our setup is like below: > SIP UAC--------->Kamailio+RTPProxy------>SIP Server------->Phone > Now when we call from UAC to Phone then voice is coming good at the > UAC side. But we receive the bad voice at the Phone side the Voice is > breaking and hardly audible. > Can anybody faced this problem. Please suggest me any solution if any. > I can mail any settings if anybody asks. > Thanks in advance. > Regards > Varun > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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How can I implement Asterisk as Media server?
by Patrik Kristel 02 May '14

02 May '14

Hello, I have Kamailio with websocket module and I want to connect Asterisk as media server. I'm trying to route calls from web JsSIP users to non-web users I would like to ask how can I implement it? Can i use the rtpproxy-ng Module and here setup IP of Asterisk? Or is there any other way to do it? Thank you for your help! Regards, Patrik Kristel

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BLF issue with Kamailio behind NAT
by Errol Samuels 02 May '14

02 May '14

Hello All, I have an integration with Kamailio and FusionPBX (Freeswitch) working great but I been trying to troubleshoot a BLF issue with Kamailio where the BLF seems to only work when Kamailio is on a Public IP rather than behind NAT. The BLF works fine when we register directly with Freeswitch either on a Public IP or behind NAT. Both Kamailio and FusionPBX (Freeswitch) are behind pfSense firewall on private IPs and registrations, Inbound and Outbound Calls etc are working fine. Just this BLF issue giving me some grief. The phone in question is a Yealink T20. I made some SIP Traces and I noticed the following: In the FS Message Header I see we have the SUBSCRIBE: ***** Begin Freeswitch trace**** Frame 13: 619 bytes on wire (4952 bits), 619 bytes captured (4952 bits) Internet Protocol Version 4, Src: 192.168.1.138 (192.168.1.138), Dst: 207.xx.xx.109 (207.xx.xx.109) User Datagram Protocol, Src Port: x11 (6062), Dst Port: ca-2 (5065) Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:testfs.mydomain.com:5065 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.138:6062;branch=z9hG4bK748715490 From: "302" <sip:302@testfs.mydomain.com:5065>;tag=530545144 To: "302" <sip:302@testfs.mydomain.com:5065> Call-ID: 2118669441(a)192.168.1.138 CSeq: 8 REGISTER Contact: <sip:302@192.168.1.138:6062> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T20P 9.72.0.30 Expires: 300 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 Frame 14: 698 bytes on wire (5584 bits), 698 bytes captured (5584 bits) Internet Protocol Version 4, Src: 207.xx.xx.109 (207.xx.xx.109), Dst: 192.168.1.138 (192.168.1.138) User Datagram Protocol, Src Port: ca-2 (5065), Dst Port: x11 (6062) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.138:6062 ;branch=z9hG4bK748715490;received=2.xx.xx.187;rport=6062 From: "302" <sip:302@testfs.mydomain.com:5065>;tag=530545144 To: "302" <sip:302@testfs.mydomain.com:5065>;tag=j71FeN5gcBjDe Call-ID: 2118669441(a)192.168.1.138 CSeq: 8 REGISTER Contact: <sip:302@192.168.1.138:6062 ;received=2.xx.xx.187:6062>;expires=300 Date: Thu, 24 Apr 2014 17:03:18 GMT User-Agent: FreeSWITCH-mod_sofia/1.2.23+git~20140419T040926Z~8a20da75ef~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 Frame 15: 509 bytes on wire (4072 bits), 509 bytes captured (4072 bits) Internet Protocol Version 4, Src: 192.168.1.138 (192.168.1.138), Dst: 10.22.110.9 (10.22.110.9) User Datagram Protocol, Src Port: x11 (6062), Dst Port: ca-2 (5065) Session Initiation Protocol (SUBSCRIBE) Request-Line: SUBSCRIBE sip:301@10.22.110.9:5065 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.138:6062;branch=z9hG4bK148600064 From: "302" <sip:302@testfs.mydomain.com:5065>;tag=2146254460 To: <sip:301@testfs.mydomain.com:5065>;tag=70uoBUVRvJyr Call-ID: 168472766(a)192.168.1.138 CSeq: 2 SUBSCRIBE Contact: <sip:302@192.168.1.138:6062> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: Yealink SIP-T20P 9.72.0.30 Expires: 1800 Event: dialog Content-Length: 0 ****** End Freeswitch trace ******** In the Kamailio Header there is no SUBSCRIBE for the BLF. ***** Begin Kamailio trace**** Frame 17: 809 bytes on wire (6472 bits), 809 bytes captured (6472 bits) Internet Protocol Version 4, Src: 192.168.1.138 (192.168.1.138), Dst: 207.xx.xx.108 (207.xx.xx.108) User Datagram Protocol, Src Port: x11 (6062), Dst Port: sip (5060) Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:test.mydomain.com:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.138:6062;branch=z9hG4bK1572502835 From: "302" <sip:302@test.mydomain.com>;tag=1877226231 To: "302" <sip:302@test.mydomain.com> Call-ID: 887242692(a)192.168.1.138 CSeq: 3 REGISTER Contact: <sip:302@192.168.1.138:6062> Authorization: Digest username="302", realm="test.mydomain.com", nonce="U1lHHFNZRfB7/5Lyb/Rz3p/xu26B1lpE", uri="sip:test.mydomain.com:5060", response="7687c7d3141d1898e63a1cc9538f05d7", algorithm=MD5 Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T20P 9.72.0.30 Expires: 300 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 Frame 18: 496 bytes on wire (3968 bits), 496 bytes captured (3968 bits) Internet Protocol Version 4, Src: 207.xx.xx.108 (207.xx.xx.108), Dst: 192.168.1.138 (192.168.1.138) User Datagram Protocol, Src Port: sip (5060), Dst Port: x11 (6062) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.138:6062 ;branch=z9hG4bK1572502835;rport=6062;received=2.xx.xx.187 From: "302" <sip:302@test.mydomain.com>;tag=1877226231 To: "302" <sip:302@test.mydomain.com >;tag=ff6b32c6b28b8954915484dd79f46b1c.fcdc Call-ID: 887242692(a)192.168.1.138 CSeq: 3 REGISTER Contact: <sip:302@192.168.1.138:6062 >;expires=300;received="sip:2.xx.xx.187:6062" Server: kamailio (4.0.6 (x86_64/linux)) Content-Length: 0 Frame 19: 316 bytes on wire (2528 bits), 316 bytes captured (2528 bits) Internet Protocol Version 4, Src: 207.xx.xx.108 (207.xx.xx.108), Dst: 192.168.1.138 (192.168.1.138) User Datagram Protocol, Src Port: sip (5060), Dst Port: x11 (6062) Session Initiation Protocol (OPTIONS) Request-Line: OPTIONS sip:2.xx.xx.187:6062 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.22.110.11:5060;branch=0 From: sip:pinger@kamailio.org ;tag=uloc-53579917-4e9-89-0b632cbb-a8aa99c To: sip:2.xx.xx.187:6062 Call-ID: f9749d7-2d0b0be7-c678c54(a)10.22.110.11 CSeq: 1 OPTIONS Content-Length: 0 ***** End Kamailio trace**** Anyone faced this issue or have any ideas? regards, Errol

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sr-users May 2014