- sr-users - kamailio.org

Fwd: sdp extra \r\n in multipart body
by Diego Nadares 29 Aug '17

29 Aug '17

Hi Daniel! Thanks for your reply. The rtpengine module I think it's expecting two \r\n like: ...pcma/8000\r\n \r\n --uniqueboundary .... cpc,9\r\n \r\n --uniqueboundary-- I saw the code and I modified it but Can any one cofirm if multipart has to be like ...pcma/8000\r\n \r\n --uniqueboundary Or ...pcma/8000\r\n --uniqueboundary Or any one of those? Thanks again! Diego El El lun, 21 de ago. de 2017 a las 02:33, Daniel-Constantin Mierla < miconda(a)gmail.com> escribió: > Hello, > > there is \r\n after > > a=rtpmap:8 PCMA/8000 > > because > > --uniqueBoundary-- > > is on next line. > > Maybe is the fault of the code extracting sdp from multi-part body. Open > an issue on github bug tracker not to forget about it. If nobody looks at > it, I will try to do it, but it may take few days given traveling. > > Cheers, > Daniel > > On 18.08.17 22:21, Diego Nadares wrote: > > Hi guys, > > I have the following multipart body > > Content-Type: multipart/mixed;boundary=uniqueBoundary > Remote-Party-ID: <tel:1112222;phone-context=+54> > <1112222;phone-context=+54> > Content-Length: 364 > > --uniqueBoundary > Content-Type: application/sdp > > v=0 > o=user1 53655765 2353687637 IN IP4 111.111.111.111 > s=- > c=IN IP4 111.111.111.111 > t=0 0 > m=audio 6000 RTP/AVP 8 > *a=rtpmap:8 PCMA/8000* > *--uniqueBoundary--* > Content-Type: application/gtd > Content-Disposition: signal;handling=optional > > IAM, > CPC,20 > GCI,885sdfasdf1123 > > --uniqueBoundary-- > > > As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not > an extra '\r\n'. Is this correct? Or should have it? > > The thing is that rtpengine is receiving sdp with no final '\r'n' from > kamailio: > > Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 > 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 > 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 > PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": > [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", > "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", > "111.111.111.111" ], "from-tag": "1", "command": "offer" } > > And the resulting sdp from rtpengine is this: > > --uniqueBoundary > Content-Type: application/sdp > > v=0 > o=user1 53655765 2353687637 IN IP4 172.16.213.16 > s=- > c=IN IP4 172.16.213.16 > t=0 0 > m=audio 35066 RTP/AVP 8 > *a=rtpmap:8 PCMA/8000a=sendrecv* > a=rtcp:35067 > > The problem is related with that '\r\n' but Is this a malformed message or > a possible bug? > > Thanks in advance! > > Diego > > > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > >

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Query in LCR Rule Target Table:
by Logeshwaran G 29 Aug '17

29 Aug '17

Hi, I need to know What is the Purpose of Priority and Weight column in Lcr_rule_target table: Example: | id | lcr_id | rule_id | gw_id | priority | weight | status | createdBy | createdDate | modifiedBy | modifiedDate | +----+--------+---------+-------+----------+--------+--------+-----------+-------------+------------+--------------+ | 1 | 1 | 1 | 1 | 1 | 1 | 1 | NULL | NULL | NULL | NULL | | 2 | 1 | 4 | 1 | 1 | 2 | 1 | NULL | NULL | NULL | NULL Kindly someone please Guide me. Thanks & Kind Regards, Logeshwaran G

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LCR dont_strip_or_prefix_flag
by Patrick Wakano 29 Aug '17

29 Aug '17

Hello list, I am using Kamailio 5.0.2 on centos7 and I am trying to use the dont_strip_or_prefix_flag ( https://www.kamailio.org/docs/modules/5.0.x/modules/lcr.html#idp49131644) for LCR routing but I discovered that just by declaring the flag like this the module won't prefix my calls: #!define LCR_DONT_STRIP_OR_TAG 7 modparam("lcr", "dont_strip_or_prefix_flag", LCR_DONT_STRIP_OR_TAG) It doesn't matter if I set or reset the flag it never prefix the ReqUR if the declaration is present. I have to remove the declaration to make the module prefix it correctly. But then the whole purpose of the flag is lost....Anyone aware of this behaviour? Regards, Patrick Wakano

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TSILO + NAT issues
by Emanuele Gambaro 28 Aug '17

28 Aug '17

Hi to all. I have the following setup-needs : *) Kamailio 5.0.1 with TSILO module and Apple push notification service. *) Rtpengine configured in order to force a "media-handover” for every call. *) NAT forced for all connection. I have a lot of problem with the INVRESUME route. When an INVITE is resumed (upon mobile client connection via APN message) the NAT does not work, and there is no rtp flow between devices. It works only if device is already online (so without the SUSPEND route). How can I setup TSILO in order to force NAT management for all calls? Thank you -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

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lcr gws
by Logeshwaran G 28 Aug '17

28 Aug '17

Hi, I am having two lcr gateways in lcr_gw table. But When executing the "kamcmd lcr.dump_gws", Its showing only one gateway list: [root@zeodialer ~]# kamcmd lcr.dump_gws { lcr_id: 1 gw_id: 1 gw_index: 1 gw_name: Carrier1 scheme: sip: ip_addr: 104.251.178.29 hostname: port: 5060 params: transport: ;transport=udp strip: 0 prefix: tag: flags: 1 state: 0 defunct_until: 0 } Did I missing something? Thanks & Kind Regards, Logeshwaran G

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Fwd: Weird issue with kamailio relaying messages to itself
by George Diamantopoulos 26 Aug '17

26 Aug '17

Stupid gmail replying to sender only... ---------- Forwarded message ---------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: 27 August 2017 at 02:29 Subject: Re: [SR-Users] Weird issue with kamailio relaying messages to itself To: Daniel-Constantin Mierla <miconda(a)gmail.com> Hello all, I've figured out what was going on, so I'm sharing here in case anyone else runs into this. I guess it should be a fairly common situation under certain circumstances when using the dispatcher module... The problem was that no destinations in the dispatcher set used were available for these requests. So $du was not set by ds_select_next. Which meant that when t_relay() was called later in the script, it would route based on R-URI, and the RURI's uri was kamailio itself. As to the reason why I was left with no dispatcher destinations available, well, I would mark destinations offline for 500 "server error" responses coming from them, and asterisk (which is the receiving application for all of dispatcher's destination sets) will send out a 500 to the B-leg when it receives a 480 to an A-leg. Getting a single 480 to one of the asterisk boxes would cause this to happen across all destinations, as kamailio would retry the next destination after a 500 failure and would receive a 500 from all of them in the end (because all of them would get the 480 in such small time frame). BR, George On 31 July 2017 at 21:52, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello Daniel, > > Thanks for the reply. That is correct, looping was not my intention, I'm > trying to figure out what's causing it... > > BR, > George > > On 31 July 2017 at 17:29, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> usually you should not loop requests locally, unless a very special case >> -- do you do the loop routing because of a need or just happens but you >> don't know why? >> >> Cheers, >> Daniel >> >> On 31.07.17 13:46, George Diamantopoulos wrote: >> >> Hello all, >> >> I have been toying with kamailio lately, and I thought I had gotten to >> the point where I had a mostly working (tm) prototype. I gave it a test >> drive with some real calls, however, and an issue manifested at least once, >> where homer received packets originating from the kamailio host, and whose >> destination was also the kamailio host. >> >> The dialog this manifested in is a generally problematic one, with many >> retransmissions occurring because of slow database access (I haven't >> implemented htable caching yet). No packet capture over the network is >> actually taking place, I'm copying everything to homer with "trace_mode"set >> to 1. >> >> Homer shows these messages like in the screenshot: >> https://imagebin.ca/v/3VGJAmovRBmo. Here's an example of a packet: >> >> >> 2017-07-28 14:32:29 +0300 : 2.3.4.5:5060 -> 2.3.4.5:5060 >> INVITE sip:1234567890@kamailio-server.org SIP/2.0 >> Record-Route: <sip:2.3.4.5;lr;ftag=as491cec82> >> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bK0cf.0779 >> 9e3eb71f33a9ef91178ac760ebd2.0 >> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bKsr-BnCVy >> rWoUladI14pU7Pry-Pzy-ONy-VSQ74NU7HzeYazq2nFU2Oc5FIWiGIKq-HXe >> x1oONpam-C6IrIXkr4FQxZRh7sM >> Max-Forwards: 69 >> From: <sip:subscriber@5.4.3.2:5061>;tag=as491cec82 >> To: <sip:1234567890@kamailio-server.org> >> Contact: <sip:2.3.4.5;line=sr-eNC05xhKefQnON1EglFrOkF0OfpzV2s9UzM9UXM >> NQXMn5-B0Q-s*> >> Call-ID: 4e5d22c3369704b97d866c8d0f3798f8@192.168.201.2:5061 >> CSeq: 103 INVITE >> User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1 >> Date: Fri, 28 Jul 2017 11:32:24 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Remote-Party-ID: "9876543210" <sip:9876543210@192.168.201.2> >> ;party=calling;privacy=off;screen=yes >> Content-Type: application/sdp >> Content-Length: 500 >> >> v=0 >> o=root 242468242 242468243 IN IP4 172.17.130.13 >> s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1 >> c=IN IP4 172.17.130.13 >> t=0 0 >> m=audio 59426 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:50 >> a=sendrecv >> a=rtcp:59427 >> a=ice-ufrag:hffIoy6x >> a=ice-pwd:nXB8ip7Qz8ZG5yyZRr97kGJbej >> a=candidate:igct4zWHEzhGCjWc 1 UDP 2130706431 <21%203070%206431> >> 172.17.130.13 59426 typ host >> a=candidate:igct4zWHEzhGCjWc 2 UDP 2130706430 <21%203070%206430> >> 172.17.130.13 59427 typ host >> >> Can something like this be triggered by misconfiguration in the routing >> scripts? Should it worry me and should I dig into it more? Could it be a >> bug of the siptrace module and nothing bad actually took place? I'm not >> sure where to start with this, so any input would be greatly appreciated. >> Thanks! >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - www.asipto.com >> Kamailio World Conference - www.kamailioworld.com >> >> >

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Kamailio fails to process ACK to 487 from Cisco-SIPGateway/IOS-12.x?
by George Diamantopoulos 26 Aug '17

26 Aug '17

Hello all, I'm having a weird issue with Kamailio failing to properly process an ACK received to a 487 it sent, resulting in retransmissions of the 487. I assume it's because it can't match the ACK to the transaction, but I could be wrong. I'm using a modified version of the default configuration, so ACKs should be handled correctly. I haven't editted the WITHINDLG route in any way that would affect this (or at least I think). In addition, ACKs to 487 from other UAs are processed correctly, and these transactions are handled by the same routes in kamailio configuration as the problematic one, so I'm inclined to believe it's UA-specific? Here's an example transaction of the failed kind (results in kamailio retransmitting the 487): myself:5060 -> peer:5060 ------------------------- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP peer:5060 From: <sip:user@peer>;tag=116B5368-24D8 To: <sip:tel@myself>;tag=as655f6372 Call-ID: 84DC69F2-873811E7-8A639B5A-3D9194E8@peer CSeq: 101 INVITE Server: modCOM v2 SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 peer:49590 -> myself:5060 ------------------------- ACK sip:tel@myself:5060 SIP/2.0 Via: SIP/2.0/UDP peer:5060 From: <sip:user@peer>;tag=116B5368-24D8 To: <sip:tel@myself>;tag=as655f6372 Date: Wed, 23 Aug 2017 12:50:47 GMT Call-ID: 84DC69F2-873811E7-8A639B5A-3D9194E8@peer Max-Forwards: 10 Content-Length: 0 CSeq: 101 ACK And here's another similar transaction which is successful (no retransmissions): myself:5060 -> peer:5060 ------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP peer:5060;branch=z9hG4bKjbmvq4009gthskk0a6s1.1 From: <sip:user@anonymous.invalid;user=phone>;tag=599D7495-9ACE9E3-0A324A05 To: <sip:tel@anonymous.invalid:5060;user=phone>;tag=as65375e5d Call-ID: 599D7495-007A5832@fath3pcu238 CSeq: 1 INVITE Server: modCOM v2 SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 peer:5060 -> myself:5060 ------------------------ ACK sip:tel@myself:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP peer:5060;branch=z9hG4bKjbmvq4009gthskk0a6s1.1 From: <sip:user@anonymous.invalid;user=phone>;tag=599D7495-9ACE9E3-0A324A05 To: <sip:tel@anonymous.invalid:5060;user=phone>;tag=as65375e5d Call-ID: 599D7495-007A5832@fath3pcu238 Max-Forwards: 69 Content-Length: 0 CSeq: 1 ACK I can't pinpoint anything wrong with the first exchange, other than the fact that for some reason, the "less than" (<) sign in the from and to URIs is escaped as &lt in homer's GUI (which also breaks CSS rendering in Firefox, I had to clear this code out). However, these escaping characters are not visible with sngrep when capturing traffic normally, and neither when doing a select in homer's database directly, so I guess it's a rendering bug in homer-ui and can be ignored (unless someone has reason to believe otherwise). Now the relevant portion of the debug log is: Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) exec: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=223 a=24 n=t_check_trans Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:1001]: t_check_msg(): msg id=104 global id=103 T start=(nil) Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:459]: t_lookup_request(): start searching: hash=54992, isACK=1 Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:494]: t_lookup_request(): proceeding to pre-RFC3261 transaction matching Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:641]: t_lookup_request(): no transaction found Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:1070]: t_check_msg(): msg id=104 global id=104 T end=(nil) Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) exec: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=231 a=2 n=exit It explicitly states that no transaction is found, after initiating pre-RFC3261 (why?) transaction matching. However, even pre-3261 matching should work, as the from and to headers as well as call-id in request and repy are the same. Any input would be greatly appreciated, thanks! George

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One Kamailio and many RTPENGINEs
by Denys Pozniak 25 Aug '17

25 Aug '17

Hello! I want to build distributed topology with one KAMAILIO and 5x RTPENGINEs for each region. I will have dedicated links between these sites, so I can garantee somehow voice quality. If one customer from region1 calls customer region2, RTP should go: Customer1->RTEPNGINE1->RTPENGINE2->Customer2 So the main question: Can one Kamailio build "call" between two RTPENGINEs ? BR, Denys

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Handeling Proxy Cancel event
by Abdul Basit 25 Aug '17

25 Aug '17

Hi, I am trying to capture and distinguish two types of cancels generated by kamailio. 1- Cancel when Invite timeout 408 2- Cancel generated when call lands on multiple devices and any one of them answer the call. This type of cancel comes with (rfc3326) Reason: SIP;cause=200;text="Call completed elsewhere" How can I capture and isolate these proxy generated CANCEL events? -- regards, abdul basit

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Diaplan module
by Nelson Migliaro 24 Aug '17

24 Aug '17

Hello, I am trying to use the dialplan module but I am having this problem: First of all, I pull the "dpid" value from the two first digits of the dnis: $var(dialplan_vendor) = $(rU{s.substr,0,2}); Then if(!dp_translate("$var(dialplan_vendor)","$dlg_var(servicio)/$var(ani_dialplan)")) { xlog("L_NOTICE","Error al seleccionar el dialplan\n"); sl_send_reply("403", "Not allowed"); drop; } But when I test everything I get: ERROR: dialplan [dialplan.c:233]: dp_get_ivalue(): no AVP, XAVP or SCRIPTVAR found (error in scripts) I think the problem is related to variable $var(dialplan_vendor). It is set up as string due to the transformation and dialplan need a integer. If I set up the variable this way for testing: $var(dialplan_vendor) = 18, it works correctly. Question, how can I transform the sting into a integer? Thank you Nelson.

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Consistency: -1 sometimes true, sometimes false
by Sebastian Damm 24 Aug '17

24 Aug '17

Hi, we recently stumbled upon the following problem: We had something like this in our route: $var(ret) = allow_address("2"); if ($var(ret)) { do_something; } Turns out, this always evaluates to true, no matter whether the IP address really is in address group 2 or not. The values in $var(ret) are 1 or -1, depending on the outcome of allow_address(). However, if we write it this way if (allow_address("2")) { do_something; } the code inside is only executed if the IP address is in address group 2. Is this some kind of inconsistency? Should negative values in variables evaluate to true or false? Best Regards, Sebastian

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Feature request: USRLOC jsonrpc ul.add - set all entry fields
by Dmitry Sytchev 23 Aug '17

23 Aug '17

Hi Daniel! Looking into code it seems RPC functions of USRLOC don't allow setting all fields for USRLOC entry, for example, Received and Socket. It would be great if we can set these, because we need to have ability to transfer contacts from other server to Kamailio. I think at present this can be done sending specially crafted fake REGISTER packets or calling special route from HTTP with some tricks for saving contact and received, but as for me it would be great to set it from RPC because we already have ul.add method. It also would be cool if we can use named parameters in JSONRPC params: {} section with names corresponding to usrloc fields, like 'table', 'AOR', 'received' and so on. Maybe on specual key not to mix with current 'params'. Thanks for your great work! -- Best regards, Dmitry Sytchev, IT Engineer

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Feature request: USRLOC jsonrpc ul.add - set all entry fields
by Dmitry Sytchev 23 Aug '17

23 Aug '17

Hi Daniel! Looking into code it seems RPC functions of USRLOC don't allow setting all fields for USRLOC entry, for example, Received and Socket. It would be great if we can set these, because we need to have ability to transfer contacts from other server to Kamailio. I think at present this can be done sending specially crafted fake REGISTER packets or calling special route from HTTP with some tricks for saving contact and received, but as for me it would be great to set it from RPC because we already have ul.add method. It also would be cool if we can use named parameters in JSONRPC params: {} section with names corresponding to usrloc fields, like 'table', 'AOR', 'received' and so on. Maybe on specual key not to mix with current 'params'. Thanks for your great work! -- Best regards, Dmitry Sytchev, IT Engineer

1 0

Kamailio APP_LUA
by santosh gupta 23 Aug '17

23 Aug '17

Hi Robert, Hope you are doing good. You have helped me in the past. I need to set LCR via app_lua, getting rates and destination from mysql. Link which i am following is https://www.kamailio.org/wiki/embeddedapi/4.0.x/lua The issue is I am not able to get connect via sr commands for example sr.sqlops.query("connection", "query", "result") What should be exactly is "connection", "query", "result" Error I am getting in syslog is /usr/local/kamailio/etc/kamailio/kamailio.lua:1: attempt to index field 'sqlops' (a nil value) I am using kamailio 5.0.2 A little bit of headsup will help me. Thanks a lot again. -- Regards Santosh Gupta +91 9323043570 <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Query in Dialplan module
by Logeshwaran G 23 Aug '17

23 Aug '17

Hi, I have one query in Dialplan module, my scenario is If I call 22291+++++++++ the first three digit should remove and the call place to 91+++++++. But i dont get the exact result, the first three digits are not removing. I have used the below rules: Lcr_rule table: ============ id: 5 lcr_id: 2 Prefix:22291 Request uri: NULL Stopper: 1 Enabled: 1 LCR_rule_target: ================== id:3 lcr_id:2 rule_id:5 gw:id:2 priority:1 weight:1 Dialplan table: ============= id:3 dpid:2 pr:1 match_op:1 match_exp: ^222[0-9]+$ subst_exp: ^222[0-9]+$ repl_exp:\1 Is it correct or not, Please guide me. Thanks & Kind Regards, Logeshwaran G

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Set up caller ID
by Nelson Migliaro 23 Aug '17

23 Aug '17

Hello, I have several Asterisk sending calls to a centralizad Kamailio. I would like this Kamailio to set up the caller id number of the calls. I am using the uac_replace_from function but I would like to pull the caller id from a database to set up the number based on the sip vendor the call is forwarded What would be the correct way to do that? I was thinking of: sql_query("ca", "select ddi from ddi_table where xxx= $rU limit 1", "ra"); $avp(dnis) = $dbr(ra=>[0,0]); sql_result_free("ra"); But I am afraid of causing a bottheneck on the database. As usual, thank you for the help. Regards, Nelson.-

2 1

rtpproxy rtp timeout
by yu＠yu-boot.ru 23 Aug '17

23 Aug '17

Hello. What version of rtpproxy app should be used for this feature to work with Kami? How to use it correctly? -- Отправлено из myMail для Android

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snmpstats get calls that have ended in 5xx cause code
by Olli Attila 23 Aug '17

23 Aug '17

Hello, I have Kamailio 5.0 configured with snmpstats module. I can succesfully query for eg. the current num of dialogs on kamailio by issuing: snmpget -v 2c -c public localhost KAMAILIO-MIB::kamailioCurNumDialogs.0 KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 Is it possible to make query about calls that have ended with 5xx sip cause code on last 5min or 15min? I noticed that the following is available on the snmpstats module but it only holds a counter that keeps increasing until I restart kamailio process: snmpget -v 2c -c public localhost KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 Also I think the above includes all cause codes that are between 3xx-6xx. Cheers, --Olli

1 0

cr reload in xmlrpc
by Logeshwaran G 23 Aug '17

23 Aug '17

Hi, It is the correct way of using the cr reload command in xmlrpc through postman? <?xml version='1.0'?> <methodCall> *<methodName>cr.reload</methodName>* </methodCall> Thanks & Kind Regards, Logeshwaran G

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Save original $fU, $rU
by yu＠yu-boot.ru 22 Aug '17

22 Aug '17

Have a problem with Asterisk. When callee behind Asterisk hangs up first, it sends back BYE with swapped From and To fields, which affects Kami CDRs rendering them useless. I've tried to save original From/To to AVPs on initial INVITE but they are empty when CDR is written. How to solve this? -- Отправлено из myMail для Android

3 2

dp_translate_cr_route, R-URI
by Yu Boot 22 Aug '17

22 Aug '17

Hello. #cut off prefix "9" dp_translate("1"); ... #searching in db by called number without prefix... if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ ... and failing. Is it normal behavior that cr_route "sees" original $rU even after dp_translate manipulations?

1 0

Kamailio / Asterisk and multidomain
by Cyrille Demaret 22 Aug '17

22 Aug '17

Hi, I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration" tutorials, I would like to let Kamailio handle registrations, calls between users and other basic functionalities. Asterisk will only handle advanced features like voicemail, advanced dialplan configuration,. My problem is that It needs to be multidomain. My network will be as follow with 2 network interfaces for Kamailio and a private lan between Kamailio and Asterisk : Public interface ip 1.2.3.4 ----> [Kamailio] <----- Private interface 192.168.100.10 <----------------> 192.168.100.11 [Asterisk] When I send a call to asterisk, the domain is sent in the from field of the INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP domain using the ${SIPDOMAIN} variable). My problem is when I need to send a call back to Kamailio for example to reach another user of the domain. I'm using Asterisk 14 with PJSIP with the following config : [kamailio] type=endpoint transport=transport-udp context=from-kamailio disallow=all allow=ulaw aors=kamailio [kamailio] type=aor contact=sip:192.168.100.10:5060 [kamailio] type=identify endpoint=kamailio match=192.168.100.10 If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio /sip:200@testdomain.com", Asterisk contact directly testcomain.com without going through the local IP of my Kamailio. I can't send a domain to Kamailio in the INVITE request. Does anyone can help me on this or maybe simply tell me that I'm not going to the good direction? :) Thank you, Cyrille

3 3

Siremis - can't add carrier without prefix/suffix
by Yu Boot 22 Aug '17

22 Aug '17

Hello Suddenly revealed that I can't add carriers without rewrite_prefix/rewrite_suffix via Siremis. But I don't need these fields! And kamctl adds these rules fine. What's a problem? Screenshot with error http://tinypic.com/r/68v493/9

2 4

problems with use_dst_blacklist and ICMP "Destination Unreachable"
by Valentin Christoph 22 Aug '17

22 Aug '17

Hi all, Question: * Does the TM destination blacklist work, if destination host responds with ICMP "Destination Unreachable"? Discussion: I have got an IMS scenario, where I think the destination blacklist is not working and try to find the reason. I use UDP between two Kamailio SIP Proxies (P-CSCF and S-CSCF) and send a SIP request to the primary instance of the S-CSCF. However the destination Kamailio has been stopped with kamctl stop before and so the destination machine responds with ICMP "Destination Unreachable". DNS Failover towards secondary S-CSCF instance works perfectly. Destination Blacklist does not work. I.e. the next SIP Request is sent to the primary host again, before failover to secondary host applies. I would have expected the request would be sent to secondary host immediately. >From the description in doc/dst_blacklist.txt [...] A destination is added to the blacklist when an attempt to send to it fails (e.g. timeout while trying to send or connect on TCP), or when a SIP timeout occurs while trying to forward statefully an INVITE (using tm) and the remote side doesn't send back any response.[...] I think the destination blacklist does not work, because an ICMP Response "Destination Unreachable" is received. Second question: * How could I avoid the problem? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

rtpproxy: One direction rtp not being relayed
by Jon Bonilla (Manwe) 22 Aug '17

22 Aug '17

Hi all After a migration, a customer is reporting one way audio on incoming calls. A tcpdump capture shows 65 rtp packets in both directions but rtpproxy log shows RTP stats: 1 in from callee, 64 in from caller, 65 relayed, 0 dropped Any thoughts? cheers, Jon

2 1

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