- sr-users - kamailio.org

kamailio with kannel
by eyas barhouk 02 Jul '18

02 Jul '18

hello dears , i'm trying to use "http_client" module to post SMS from kamailio to kannel using the following configuration : ....... modparam("http_client", "httpcon", "kannel=>http://username:password@127.0.0.1:443") .......... ......... http_connect("kannel","cgi-bin/sendsms?user=username&pass=password&charset=utf-16&coding=2&from=$avp(from)&to=$avp(to)&text=$(avp(text){s.escape.user})","$var(result)"); ....... And this test success for English characters & phrases, but when i'm trying to send Arabic or Chinese characters it only success if the content of the SMS only has one word. in other-word i mean when i'm trying to send more than one Arabic word with a space between each word it reply with HTTP error code 400. so do you have any ideas about this problem ??

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Using Kamailio+rtproxy with freeswitch
by Philippe,Edson H 02 Jul '18

02 Jul '18

Hello. I have been trying to use kamailio+rtpproxy to handle freeswitch. So far, I am able to handle the sip routing via Kamailio, however, I am not able to get the RTP portion to go through my kamailio server. it seems once Kamailio handle the initial sip, there is a 1 to 1 communication between freeswitch and my client (phone) Has anyone solved this problem?

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subst query
by mahesh b 02 Jul '18

02 Jul '18

Hi , Can someone explain me what is \1 and \3 doing in the subst expression ? Input : From: <sip:anonymous@10.211.160.168 >;tag=41008079_nab_FFFF_isp_FFFF_cco_FFFF_igo_FFFF_mgt_78DD subst('/^From:(.*)10.211.160.168(.*)>(.*)/From:\110.211.160.174>\3/ig'); Output : From: <sip:anonymous@10.211.160.174 >;tag=41008079_nab_FFFF_isp_FFFF_cco_FFFF_igo_FFFF_mgt_78DD Regards, Mahesh.b

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Should I ignore Route header in ACK?
by Yuriy Gorlichenko 02 Jul '18

02 Jul '18

Hi I know that this is not question too much close to the kamialio users but mostly losed to the RFC specifiacations but this community looks like pretty much close to it that is why I want to ask this question here, that's why sorry and thanks for help in this question: I have a situation when provider sends me 200 response with Request-Route header and changed contact header: Means response comes from 1.1.1.1:5060 Request-Route contains: 1.1.1.1:5060 But Contact contains: 1.1.1.1:5061 My ACK (handled by kamailio) goes to the 1.1.1.1:5060 as it setted up at the Route Hedaer of ACK (because of Request-Route) but provider says me that i should use Contact for the ACK I was surprised because of https://tools.ietf.org/html/rfc3261#section-12.2.1.1 and https://tools.ietf.org/html/rfc3261#section-8.1.2 Says that I should use Route header for reaching destination But I was surprised second time when tested this scenario with FreeSwitch and another softphone (as UA) because of it both sends ACK to the based on Contact address and ignores Route I just wanna ask if I missed some scenario in the RFC when it is described to ignore Route header for the UA (I know that I use kamailio on my case as proxy server but should understand finally who should make changes with packet handling) Thanks one more time for the resonses and sorry one more time for the goal of this question that belongs to the kamailio just partially

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Is it possible to play earlmedia with kamailio
by vinod mn 02 Jul '18

02 Jul '18

Hi, I am trying to configure kamailio to play early media and i am using rtpproxy_stream2uac(), but the audio file plays only after the call is connected. How to use rtpproxy_stream2uac() to play only the early media. Thank you vinod.M.N

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Kamailio installation problem
by Dipayan Sanyal 01 Jul '18

01 Jul '18

Hi, I have installed kamailio in my system but while starting I am geeting following errors- Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: usrloc [dlist.c:644]: register_udomain(): failed to open database connection Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: registrar [registrar.c:651]: domain_fixup(): failed to register domain Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/kamailio-5.1/etc/kamailio/kamailio.cfg:643 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/rvalue.c:3859]: fix_rval_expr(): failure in cfg at line: 643 col: 22 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/rvalue.c:3859]: fix_rval_expr(): failure in cfg at line: 643 col: 22 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/kamailio-5.1/etc/kamailio/kamailio.cfg:646 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:85]: tm_shutdown(): start Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:88]: tm_shutdown(): emptying hash table Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:90]: tm_shutdown(): removing semaphores Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:92]: tm_shutdown(): destroying tmcb lists Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:95]: tm_shutdown(): done Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: <core> [core/mem/shm.c:260]: shm_destroy_manager(): destroying memory manager: q_malloc Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: <core> [core/mem/q_malloc.c:1169]: qm_shm_lock_destroy(): destroying the shared memory lock Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: <core> [core/mem/pkg.c:90]: pkg_destroy_manager(): destroying memory manager: q_malloc Please look into the problem. Thanking You, Dip

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Fwd: Video call recording and mobile push notifications
by Jeremy Renner 01 Jul '18

01 Jul '18

Hi, I would like to setup up my SIP server / PBX for my business, now we have some candidates: 1. Open source solution: - Asterisk PBX, - Freeswitch PBX - Kamailio - OpenSIPS 2. Business solution: - Brekeke PBX(https://www.brekeke.com - Vodia PBX(https://www.vodia.com) - 3CX PBX(https://www.3cx.com) - PortSIP PBX(https://www.portsip.com/portsip-pbx) *Below features are mandatory for our project:* - Video call recording (For the finance industry, the video recording is necessary) - Push notifications for mobile app - Multi-tenant support - Both Linux and Windows support (at 1st stage, we would like to run it on Windows server and migrate it to Linux server in the future if users increased), the Linux support is required, the Windows support is preferred. We have some questions: 1. Does the the Kamailio can works as a PBX ? 2. If yes, does the Kamailio support push notifications and video recording ? 3. Does the Kamailio can works for Multi-tenant ? 4. Does Kamailio support Windows ? So far according to our research, with the business solution: - The Vodia PBX, PortSIP PBX and brrekeke all are support Multi-tenant, the 3CX is not. - The 3CX and PortSIP support push notifications, - The PortSIP also provide client SDK, with 3CX we only see the 3CX provide client apps, does 3CX has client SDK provided ? - It's seems all these PBX are support video recording ? - The PortSIP PBX and 3CX both support Linux. Please help me to make the decision, base on your experiences, which one (open source or business solution) is good to us ? I'm really new to VoIP... Thanks in advance. Best regards,

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linphone & presence_xml ERROR
by Pali Rohár 01 Jul '18

01 Jul '18

Hi! I'm using linphone 3.6.1 as SIP client and Kamailio 4.4.4 as proxy server and registrar. And every time I try to change online status (presence) in linphone client, I see following error messages in kamailio server log: ERROR: presence_xml [add_events.c:167]: xml_publ_handl(): bad body format ERROR: presence [publish.c:443]: handle_publish(): in event specific publish handling ERROR: tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 500 reply when a final 415 was sent out ERROR: sl [sl.c:269]: send_reply(): failed to reply stateful (tm) ERROR: presence [utils_func.c:146]: send_error_reply(): sending 500 Server Internal Error reply ERROR: presence [publish.c:492]: handle_publish(): failed to send error reply So it looks like kamailio is not able to parse presence change which is linphone doing. I captured traffic via tcpdump what linphone sends: 21:58:58.704792 IP (tos 0x68, ttl 64, id 64109, offset 0, flags [DF], proto UDP (17), length 32) LINPHONE_IP_ADDRESS.5060 > KAMAILIO_IP_ADDRESS.5060: SIP 21:58:58.704919 IP (tos 0x68, ttl 64, id 64110, offset 0, flags [DF], proto UDP (17), length 752) LINPHONE_IP_ADDRESS.5060 > KAMAILIO_IP_ADDRESS.5060: SIP, length: 724 PUBLISH sip:USER@HOST SIP/2.0 Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport;branch=z9hG4bK2049418743 From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST> Call-ID: 1435474953 CSeq: 26 PUBLISH Content-Type: application/pidf+xml Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Expires: 600 Event: presence Content-Length: 353 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="User <sip:USER@HOST>"> <tuple id="sg89ae"> <status><basic>closed</basic></status> <contact priority="0.8">User <sip:USER@HOST></contact> </tuple> </presence> 21:58:58.710097 IP (tos 0x10, ttl 53, id 10255, offset 0, flags [none], proto UDP (17), length 473) KAMAILIO_IP_ADDRESS.5060 > LINPHONE_IP_ADDRESS.5060: SIP, length: 445 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport=5060;branch=z9hG4bK2049418743;received=EXTERNAL_IP_ADDRESS From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST>;tag=515cb47c4c34ffa85b598d6b25676122.744d Call-ID: 1435474953 CSeq: 26 PUBLISH Proxy-Authenticate: Digest realm="HOST", nonce="NONCE" Server: kamailio (4.4.4 (x86_64/linux)) Content-Length: 0 21:58:58.718147 IP (tos 0x68, ttl 64, id 64111, offset 0, flags [DF], proto UDP (17), length 940) LINPHONE_IP_ADDRESS.5060 > KAMAILIO_IP_ADDRESS.5060: SIP, length: 912 PUBLISH sip:USER@HOST SIP/2.0 Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport;branch=z9hG4bK1165141043 From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST> Call-ID: 1435474953 CSeq: 27 PUBLISH Proxy-Authorization: Digest username="USER", realm="HOST", nonce="NONCE", uri="sip:USER@HOST", response="RESP", algorithm=MD5 Content-Type: application/pidf+xml Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Expires: 600 Event: presence Content-Length: 353 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="User <sip:USER@HOST>"> <tuple id="sg89ae"> <status><basic>closed</basic></status> <contact priority="0.8">User <sip:USER@HOST></contact> </tuple> </presence> 21:58:58.781668 IP (tos 0x10, ttl 53, id 10259, offset 0, flags [none], proto UDP (17), length 380) KAMAILIO_IP_ADDRESS.5060 > LINPHONE_IP_ADDRESS.5060: SIP, length: 352 SIP/2.0 415 Unsupported media type Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport=5060;branch=z9hG4bK1165141043;received=EXTERNAL_IP_ADDRESS From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST>;tag=97d8e785fdf42bf9622a64c13c504961-3901 Call-ID: 1435474953 CSeq: 27 PUBLISH Server: kamailio (4.4.4 (x86_64/linux)) Content-Length: 0 I replaced ip addresses in packets by KAMAILIO_IP_ADDRESS, LINPHONE_IP_ADDRESS and EXTERNAL_IP_ADDRESS strings and also SIP account by USER@HOST. Maybe it helps you. Any idea why kamailio refuse presence update and reports those error into error log? Or is there any special setting which is needed for linphone or other SIP clients for online status / presence support? -- Pali Rohár pali.rohar(a)gmail.com

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dovecot authentication
by Pali Rohár 30 Jun '18

30 Jun '18

Hi! Is there any module or configuration how to use dovecot SASL authentication in Kamailio? Dovecot exports unix socket (or tcp on localhost) for authentication service and lot of server applications can be configured to use it (e.g. postfix, prosody, ...) to have one central place of user and authentication database. It supports also strong authentication schemas like SCRAM-SHA, but also plain text password or DIGEST-MD5. In SIP is used DIGEST-MD5 therefore it could be suitable also for SIP and Kamailio. -- Pali Rohár pali.rohar(a)gmail.com

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Understanding 5XX stats in Kamailio
by Joel Serrano 29 Jun '18

29 Jun '18

Hi, Does anyone know the difference between these metrics?? core:rcv_replies_5xx sl:5xx_replies tmx:5xx_transactions Specially between core and tmx? Thanks, Joel.

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Relevant development links
by Amauri Oliveira 28 Jun '18

28 Jun '18

Hello. I'm new to Kamailio perhaps I've seen it in actions at AirBNB. I'm starting the journey as a new developer who wants to grab Kamailio and build something. I would like to check if you could provide the most relevant links for a beginner also with some samples. I noticed the abundance of modules perhaps I would like to see what people are doing with it. Best Regards, Amauri.

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Fwd: Understanding is_first_hop() and how it handles in-dialog replies
by Joel Serrano 28 Jun '18

28 Jun '18

Hi guys, I'm having an issue which I have narrowed down to is_first_hop(), I can apply a workaround, but I don't know if I'm doing it correctly or if my problem is caused by misconfiguration of anything. Let's say we have the following flow: Client (NAT) TLS -> Kamailio1 (only Public IP) UDP -> Kamailio2 (only Public IP) UDP -> FreeSWITCH (only Public IP). Regarding this setup, let's focus only on Kamailio1. And for the sake of a clear example, I have done a little draw.io diagram: >From the above flow, let's stick to the *200 OK* that Kam1 is going to receive from Kam2. (marked in *blue* in the screenshot). First 200 OK (replying to initial INVITE): SIP/2.0 200 OK Via: SIP/2.0/UDP KAM1_PUB_IP;rport=5060;branch=z9hG4bK1e2b. e1307b519c9b5f0015343e13f35aeace.0;i=3 Via: SIP/2.0/TLS [2607:fb90:489b:6e13:85f8:596b:b86b:c831]:54744; received=172.58.17.149;branch=z9hG4bK.jmXSnh-7x;rpo =43842 Record-Route: <sip:KAM2_PUB_IP;lr=on;ftag=7vidaJ3Hw;did=36e.d441> Record-Route: <sip:KAM1_PUB_IP;r2=on;lr=on;ftag=7vidaJ3Hw;did=36e.a792; nat=yes> Record-Route: <sip:KAM1_PUB_FQDN:443;transport=tls;r2=on;lr=on; ftag=7vidaJ3Hw;did=36e.a792;nat=yes> From: "Joel Test 1" <sip:8bd2a0aba14541789bb7269800646458@MY_DOMAIN >;tag=7vidaJ3Hw To: "Joel Test 2" <sip:e78f2617b0d345d3bdb7b6780ece903c@MY_DOMAIN>;tag= vceg6N0m5ypHa Call-ID: 3ezoQGF1kp CSeq: 21 INVITE Contact: <sip:e78f2617b0d345d3bdb7b6780ece903c@FS_PUB_IP:6061;transport=udp> User-Agent: TP MEDIA 2.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 358 Remote-Party-ID: "e78f2617b0d345d3bdb7b6780ece903c" <sip: e78f2617b0d345d3bdb7b6780ece903c@MY_DOMAIN>;party=calling; privacy=off;screen=no v=0 o=TP 1529576021 1529576022 IN IP4 FS_PUB_IP s=TP c=IN IP4 FS_PUB_IP t=0 0 m=audio 25484 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1; maxplaybackrate=8000; sprop-maxcapturerate=8000 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=rtcp:25485 IN IP4 FS_PUB_IP Second 200 OK (replying to in-dialog INVITE with updated SDP): SIP/2.0 200 OK Via: SIP/2.0/UDP KAM1_PUB_IP;rport=5060;branch=z9hG4bKed2b. 0006a6a159e800129a62b4415fdd64e6.0;i=7 Via: SIP/2.0/TLS 192.168.30.63:54752;received=A.B.C.D;branch=z9hG4bK. iKO2iYIgK;rport=27819 From: "Joel Test 1" <sip:8bd2a0aba14541789bb7269800646458@MY_DOMAIN >;tag=7vidaJ3Hw To: "Joel Test 2" <sip:e78f2617b0d345d3bdb7b6780ece903c@MY_DOMAIN>;tag= vceg6N0m5ypHa Call-ID: 3ezoQGF1kp CSeq: 22 INVITE Contact: <sip:e78f2617b0d345d3bdb7b6780ece903c@FS_PUB_IP:6061;transport=udp> User-Agent: TP MEDIA 2.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 358 v=0 o=TP 1529576021 1529576022 IN IP4 FS_PUB_IP s=TP c=IN IP4 FS_PUB_IP t=0 0 m=audio 25484 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1; maxplaybackrate=8000; sprop-maxcapturerate=8000 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=rtcp:25485 IN IP4 FS_PUB_IP Now here comes the problem, I have the following in my kam1 config: route[NATMANAGE] { ... if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { fix_nated_contact(); } } } ... } So, on the first 200 OK, when it reaches that part of the config: 1- is_reply() -> OK 2- isbflagset(FLB_NATB) -> OK (because on the initial request NAT was detected blablabla....) 3- is_first_hop() -> FAIL fix_nated_contact() is NOT applied. (This is the correct and the expected behavior). Now, on the second 200 OK, again in that part of the config: 1- is_reply() -> OK 2- isbflagset(FLB_NATB) -> OK 3-is_first_hop() -> OK fix_nated_contact() is applied, thus the contact will be changed, and the client will send the ACK with incorrect information leading to another set of issues... >From the docs: https://www.kamailio.org/docs/modules/5.1.x/ modules/siputils.html#siputils.f.is_first_hop *4.30. is_first_hop()* *The function returns true if the proxy is first hop after the original sender. For incoming SIP requests, it means there is only one Via header. For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it (to avoid detecting in case of local loops). Note that it does not detect spirals, which can have the condition for replies true also in the case of additional SIP reply receival.* So going back to the examples: first 200 OK: 1- "top Record-Route URI is 'myself' -> FAIL So we we are NOT the first hop, we do nothing and forward the reply to the client. second 200 OK: 1- "top Record-Route URI is 'myself' -> No record-route headers are present, so we enter the is_first_hop() condition and modify the contact with fix_nated_contact(). Now to the real topic, I have a workaround as: ... if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (!ds_is_from_list()) { # <-- Check to see if the reply is coming from our internal servers fix_nated_contact(); } } } } ... And that is working correctly, but I would like to understand the reasons. I hope I have explained myself correctly, otherwise it's impossible to get to the point of my questions: 1- Is it correct for is_first_hop() to detect the second 200 OK as a first hop when it isn't? the behavior matches the documentation, so I don't know. If we stick to the check of the headers etc etc, it's working as described, if we stick to the concept of kam1 being actually being the first hop of that 200OK, then the check would need another condition to exclude the 200 OK (or better, the replies) of an in-dialog INVITE. 2- Do you guys consider the workaround something reasonable? Under my opinion I would like to not have to add that, but I also don't know. 3- Am I missing something super standard that avoids all of this? I'm starting to go crazy trying and comparing different things to get to understand this and I want to make sure it's not just a is_first_hop() bug? Sorry for such a long email, but I think that describing the scenario and flow was required. Any input is more than welcome! Thanks! Joel.

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Question on the new Kamailio v5.x Releases
by Ginhoux, Patrick 27 Jun '18

27 Jun '18

Hi, I always read your email regarding new stable releases. I am currently working on a project upgrade of multiples servers to the most recent OS (Windows, RedHat) and middleware. In this project, I have upgraded SIP Proxies server running on old version of Kamailio, to the 5.0.1 release (over RedHat 7.4). Effort to make that upgrade and tests have been done with success in the last months. So considering to upgrade to the 5.0.7 or 5.1.4 would be a subsequent work I need to evaluate. I could read all the release notes, but it is also a big challenge to determine if fixes are interesting for me. Based on your experience, is there particular cases, fixed in the new release I need to pay attention? Any comments are welcome. Regards Patrick GINHOUX

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Compile error on 4.4.7
by Andrew Chen 27 Jun '18

27 Jun '18

Hi all, I'm getting the following error when compiling 4.4.7 from src: mem/pkg.o: In function `pkg_init_manager': /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/pkg.c:63: undefined reference to `fm_malloc_init_pkg_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/pkg.c:68: undefined reference to `qm_malloc_init_pkg_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/pkg.c:72: undefined reference to `tlsf_malloc_init_pkg_manager' mem/shm.o: In function `shm_init_manager': /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/shm.c:239: undefined reference to `fm_malloc_init_shm_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/shm.c:244: undefined reference to `qm_malloc_init_shm_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/shm.c:248: undefined reference to `tlsf_malloc_init_shm_manager' collect2: error: ld returned 1 exit status make: *** [kamailio] Error 1 Build step 'Execute shell' marked build as failure Sending e-mails to: kivanov(a)fuze.com AChen(a)fuzebox.com Finished: FAILURE any thoughts? Thanks -- Andy Chen -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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Planning next IRC devel meeting
by Daniel-Constantin Mierla 26 Jun '18

26 Jun '18

Hello, I thought of start planning the next IRC devel meeting, it is quite some time since the last one with a busy spring for most of us. Among the topics to be approached would be the roadmap to the next major release, then discuss what else we can do for or with the project. My proposal as a date is Tuesday, June 27, at 14:00 UTC -- I created a page for it: - https://www.kamailio.org/wiki/devel/irc-meetings/2018a List yourself if you can participate or you prefer a different date, as well as add new topics that you consider they should be approached. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

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Siremis Log Information
by Waqar Hameed 26 Jun '18

26 Jun '18

Hi I am one of your user using siremis web interface for manipulating kamailio database. I just want to know if it is possible to disable auto siremis logs present in the directory siremis-4.2.0/siremis/log/debug.log. The file debug.log contains database queries along with user ids and password and I just want to stop that logs for security concerns. If it is possible please reply with some solution of it. Thanks in advance. Regards, Waqar Hameed

2 1

invalid char found in the new uri
by Kjeld Flarup 25 Jun '18

25 Jun '18

I have a system where the calls is not disconnected properly. I get this error in syslog when receiving BYE: Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:645]: restore_uri(): invalid char found in the new uri at pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob] Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:647]: restore_uri(): this can happen when URI values are altered by end points - skipping the update Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:645]: restore_uri(): invalid char found in the new uri at pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob] Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:647]: restore_uri(): this can happen when URI values are altered by end points - skipping the update Jun 24 08:24:11 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:645]: restore_uri(): invalid char found in the new uri at pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob] Jun 24 08:24:11 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:647]: restore_uri(): this can happen when URI values are altered by end points - skipping the update Kamailio receives this From header: f: <sip:004540294149@194.255.22.44>;tag=e042cfe1-40f7-410d-95fd-e5bb4f8976d7;transport=TCP;ob And passes this on, where it is obvious that '%' may give a problem f: <sip:0001@192.168.2%:>;tag=e042cfe1-40f7-410d-95fd-e5bb4f8976d7;transport=TCP;ob The RR headers send forward with INVITE looks like this: Record-Route: <sip:127.0.0.1:5069;r2=on;lr=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> Record-Route: <sip:192.168.2.9:5069;r2=on;lr=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> And received on BYE Route: <sip:127.0.0.1:5069;lr;r2=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> Route: <sip:192.168.2.9:5069;lr;r2=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> RR does not seem to be the problem as mentioned here: https://opensips.org/pipermail/users/2010-October/014999.html I have attached a pcap file with the full trace. The error appears in frame 101 The setup is a bit complex, as I have 8 Kamailio's running, where 6 is identical. The first Kamailio is on port 5069, which receives a call from a door phone. It then branches the call to registrations or GSM targets, which is handled by the other Kamailio's Thus 5069 is only doing a lookup in a database and branching, plus it handles rtpengine. I first saw the error on version 4.4, and now I replicated it on 5.1.4. My hardware is a Raspberry Pi. /usr/local/sbin/kamailio -v version: kamailio 5.1.4 (arm6/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT-NOSMP, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 23:45:36 Jun 19 2018 with gcc 4.9.2 -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 1

[topos] no local address - do record routing for all initial requests
by Abdoul Osséni 25 Jun '18

25 Jun '18

Hello all, I use topos with topos_redis modules. For each call, I got the following messages: Jun 25 09:57:37 sd-110402 /usr/local/sbin/kamailio[11867]: {1 20 ACK FKgR7X~stk x.x.x.x 24860}WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Jun 25 09:57:37 sd-110402 /usr/local/sbin/kamailio[11839]: {1 102 BYE FKgR7X~stk x.x.x.x xxxx}WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests It seems, there is no issue on the calls but I want to understand why I got these warning messages. Thank you. Abdoul

2 1

Kamailio installation
by Danny Roberts 23 Jun '18

23 Jun '18

Good Day, I have been task by my company to install Kamailio, can I get some assistance in installing the application? Also is there a GUI for easy install? DANNY ROBERTS | Solutions Technician | dtroberts(a)clearharbor.com<mailto:dtroberts@clearharbor.com> Frequente Industrial Park | P.O.BOX 1499 | St. George's, Grenada o 678-566-3212 x70778 m 473-417-5676 | www.clearharbor.com [cid:image004.png@01D3F36A.B17ABBD0] [cid:image006.png@01D3F36A.B17ABBD0] [cid:image003.png@01D406F0.8843CF10] [cid:image010.png@01D3F36A.B17ABBD0]

7 19

ims_usrloc_pcscf module hashing_type parameter does not work
by tyd 22 Jun '18

22 Jun '18

Dear sir, I'm working on Kamailio 4.4.0 and 5.1.4. When registering to Kamailio IMS using the same ip and port for 30 user (different contact user part), the pcscf get the same hash number and aor value. The hashing_type = 0 seems not to be work. Can anyone help me?

1 0

kamailio 5.1.4 for centos 7 repo
by Aidar Kamalov 22 Jun '18

22 Jun '18

Hello, http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… In file information block you can see file size is 5643596 bytes. But if you download it - file size is 5642888. And it is not possible to install kamailio 5.1.4 on centos7 from repo. -- Aydar A. Kamalov

2 2

Difference between allow trusted and allow address
by Wilkins, Steve 21 Jun '18

21 Jun '18

Hi All, Can someone explain the difference between allow trusted and allow address. Thank you!

2 1

Newbie: TOPOH in SBC/Proxy scenario regarding To-header
by Kevin Olbrich 21 Jun '18

21 Jun '18

Hi! I set up a proxy in front of an asterisk server to handle registration and similar. This setup needs topology hiding. I enabled and configured TOPOH which is working well. The only header which leaks network details is "To". It lists the proxys internal IP. Sure, a proxy should not alter the To and I am investigating if this can be changed in asterisk but is it even possible? Kind regards Kevin

3 2

Kamailio + FreeSwitch + WebRTC
by Emanuel Gianico 21 Jun '18

21 Jun '18

Hi everybody, I followed this tutorial https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamail… And it works fantastic! The next step was to add WebRTC support, so I added WebSockets module to enable web clients to register on kamailio. It works flawlessly and webphone clients register OK! (Followed this http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket ) Now, when I call from a softphone (eyeBeam) to the web client (jssip) the other party reach okay, rings okay, but when I pickup de call (from the web client), the softphone goes directly to the VoiceMail. >From the logs I see the jssip throw this error: "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." I would like to avoid RTPEngine, because from what I understand, FreeSwitch can handle the media part. Can somebody please have the kindness to guide me on how to enable webrtc between Kamailio and FreeSwitch? If somebody needs to see the "kamailio.cfg" please let me know, and i would upload the file to a gist. Cheers, Emanuel.

5 10

Kamailio v5.0.7 Released
by Daniel-Constantin Mierla 21 Jun '18

21 Jun '18

Hello, Kamailio SIP Server v5.0.7 stable release is out. This is a maintenance release of the previous stable branch, 5.0, that includes fixes since the release of v5.0.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.7 or v5.1.x. For more details about version 5.0.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/06/kamailio-v5-0-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.1, at this moment with its latest release v5.1.4. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com

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