sr-users May 2016

sr-users@lists.kamailio.org

86 participants
128 discussions

Question about testing RTPENGINE
by Dmitry 20 May '16

20 May '16

Hello I am trying to make a test setup to test the RTPENGINE. Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.132)-----via NAT-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) First WITHOUT RTPENGINE : When A calls to B - the sound both ways, so it is ok. Second WITH RTPENGINE: When A calls to B - the sound both ways, so it is ok. Actually what I want is to … [View More]

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WebRTC
by Moacir Ferreira 20 May '16

20 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes … [View More]

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Re: [SR-Users] Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 19 May '16

19 May '16

Is there is reason why this post does not get approved? From: Pejman Rajabian [mailto:pejman@italkip.com] Sent: Wednesday, May 18, 2016 12:59 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio notify conflict when a second call lands on a phone I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the … [View More]

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Sipcapture on SPARC
by Spencer Thomason 18 May '16

18 May '16

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some sore of endianness issue? Thanks for any assistance! Regards, Spencer

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[kamailio 5.0] upgrade VS rollback/backward compatibility
by Alex Lutay 18 May '16

18 May '16

Hi, We are heavily using Kamailio in our product Sipwise NGCP and currently testing possibility to provide rollback for the customers with the huge systems as the last resort functionality in case if upgrade went wrong. We hope never use it on practice but prefer to have such possibility. The main logic here is to change DB in 'backward compatible' way. So OLD code must work with NEW DB schema (but NOT a vice verse). We have some rules which helps us care about it, like "Do not drop columns, … [View More]just discontinue them" and "All new fields should have defaults", etc. Also we have separated 'code', 'configs' and 'data' parts of our product and rollback 'code'/'configs' only, while 'data' (e.g. MySQL DB) stays in new state. Otherwise rollback will require a lot if time to revert all DB statements and sometimes it is just not possible. While every time we have to upgrade Kamailio we have a problems with rollback possibility. The point here that Kamailio checks DB table_version and doesn't start if the version mismatch found. Q: What about to add Kamailio possibility to work in backward compatible mode? Sample for upgrade Kamailio 4.3.x -> 4.4.x (the versions here as an example only!) https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0 We have 3 DB statements there which are adding new columns to 3 tables, like > -- table: active_watchers > ALTER TABLE active_watchers > ADD COLUMN flags INT(11) NOT NULL DEFAULT '0', > ADD COLUMN user_agent VARCHAR(255) NOT NULL; > DELETE FROM version WHERE TABLE_NAME='active_watchers'; > INSERT INTO version (`table_name`, `table_version`) VALUES ('active_watchers','12'); Problem here is missed default for column 'user_agent' which can be empty value as for me. So, lets assume we have it there. Then we can add new column to table 'version', like that: > mysql> select * from version where table_name='active_watchers'; > +-----------------+---------------+---------------------+ > | table_name | table_version | table_compatibility | > +-----------------+---------------+---------------------+ > | active_watchers | 12 | 11 | > +-----------------+---------------+---------------------+ > 1 row in set (0.00 sec) Currently: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 Proposed behaviour: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 and compatibility option is not set - if compatibility option is set, check value of table_compatibility and refuse to load if current version less then table_compatibility value >From the sample above I would say module presence in Kamailio 4.4.x is compatible with Kamailio 4.3.x if we define default value for the field 'user_agent'. Having that will allow us to rollback on previous NGCP release and _temporary_ work on Kamailio 4.3.x having new Kamailio DB 4.4.x P.S. This is just a question to check Kamailio community vision here and reality is more complex as described, so some additional checks/limitations will be in place. P.P.S Compatibility Kamailio option can/should be disabled by default. Field 'table_compatibility' can be outside table 'version', e.g. in separate table. Complete compatibility logic can be some separate Kamailio 'compatibility' module. Thank you! -- Alexander Lutay Head of Quality Assurance Sipwise GmbH, Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Office: +43(0)13012036 Email: alutay(a)sipwise.com Website: https://www.sipwise.com Meet us @ ANGACOM: Hall 10.1 / booth N10 Exhibition and Congress for Broadband, Cable & Satellite: 07 – 09 June, 2016 in Cologne [View Less]

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Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 18 May '16

18 May '16

I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of … [View More]

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invalid route error when turning on Roles in kamailio kazoo
by Dmitry 18 May '16

18 May '16

hello When I uncomment in /etc/kazoo/kamailio/local.cfg#!trydef TRAFFIC-FILTER-ROLE (it was commented previously) And the kamailio does not start and the log gives this error:-Invalid route satement The structure of files is simple:the trafic-filter-role is in the separate file. and in the main file (default.cfg #!ifdef TRAFFIC_FILTER-ROLEinclude_file "traffic-filter-role.cfg" and that's it. But it gives INVALID ROUTE ERROR

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 17 May '16

17 May '16

Hello, David. Sorry for the time to reply you. I was a little busy here in the last few days. I am not sure about the "==". We had tried it here, but it only worked when we used with =~. We are currently running kamailio 4.2, so it may be a problem with our version. About the $hdr variable, it is used to get information from the SIP Headers. I am not sure, but I dont think that there is an header with Instance-ID, so you must try to find how can you get this ID in kamailio, as I don't know … [View More]where is it saved. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-13 12:24 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > After some time I was able to get this working here, and now I am coming > back to share my configuration with you guys (in case someone needs to get > this working). > > Here is what we have done: > > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > > xlog("L_INFO", "checking $tu registered with > User-Agent:$hdr(User-Agent)\n"); > if(reg_fetch_contacts("location", "$tu", "caller")) > { > xlog("L_INFO", "$(ulc(caller=>count)) records > found for $tu\n"); > $var(i) = 0; > while($var(i) < $(ulc(caller=>count))) > { > > xlog("L_INFO", "record $var(i) with > ruid:$(ulc(caller=>ruid)[$var(i)]) - > User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); > if > ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) > { > $var(ruid) = > $(ulc(caller=>ruid)[$var(i)]); > xlog("L_INFO", "unregistering $tu > ruid:$var(ruid)"); > unregister("location", "$tu", > "$var(ruid)"); > } > $var(i) = $var(i) + 1; > } > } > if (!save("location", "0x00")) > sl_reply_error(); > > exit; > } > } > > > There are some logs over there because we used it to see what was going on > while testing the configurations. > > > With this we can get just one register per User-Agent. > > > And again, thanks Olle for your help in this problem > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Sure it is! >> >> This is exactly what I was looking for! Actually I wasn't able to see >> that there was a way to perform this action! >> >> Thanks for your help, Olle! >> >> I will try to create this, and when I have it done, I will post my script >> here, so if someone else need the same problem I can help. >> >> Now, I have just one more question about my scenario: when I register a >> user, is there a way to create something like a "custom field" on the >> location? To explain better: If possible, I can add something like a custom >> header that says if the user is registered from a phone device or the web >> interface, then I can perform the loop and replace only the oldest contact >> using that device... >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> This is my problem... I want to have 2 contacts for AOR, and not only >>> one. >>> >>> When the 3rd arrives, it must remove the oldest and continue with 2... >>> >>> *If I am using the 0x04 parameter, it will save just one contact, and >>> the user will not be able to get the Invite both, in the mobile device and >>> web interface.* >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >>> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello, *Olle!* >>>> >>>> *Thanks for your help in this case, and don't worry about the time >>>> (actually I wasn't even expecting to get an answer today).* >>>> >>>> >>>> *I know the max_contacts parameter. But the problem is that if I set >>>> the max_contacts to 2, when the user tries to register again, he gets an >>>> error message 503. The point is that I want to get the user registered, and >>>> replace the oldest location entry for the newest one.* >>>> >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>>> >>>>> Hello all. >>>>> >>>>> I have problem here and I tried to find a solution and search over >>>>> internet, but without success. >>>>> >>>>> My scenario is the following: I have an application that must be >>>>> registered in Kamailio when a user logs in the web interface, so he can get >>>>> calls (something like a web softphone using webrtc). At this point, we are >>>>> OK, and everything is working fine. >>>>> >>>>> To get these register functions I am using the parameter >>>>> "save("location", "0x04")" as described in the REGISTRAR module >>>>> documentation, so if a user logs in another web browser or computer, only >>>>> the last one will continue registered and all calls will be forwarded to >>>>> him. >>>>> >>>>> The point is that now we are creating a phone app that will do almost >>>>> the same thing as the web interface, allowing users to receive calls using >>>>> the mobile device, and here is my problem: I want to allow my users to be >>>>> registered on two devices at the same time, but if a user logs into another >>>>> device, I don't want to reply with a 503. I want to allow the user to >>>>> register again, deregistering the oldest contact. >>>>> >>>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>>> description it says "implemented overwrite oldest contact behaviour", >>>>> but I couldn't find any documentation about it. >>>>> >>>>> So, is there a way to get this working today? >>>>> >>>>> >>>>> *Bruno Emer* >>>>> >>>>> Mobile: +55 11 96540-0044 >>>>> email: brunoemer(a)gmail.com >>>>> >>>> >>>> >>> >> > [View Less]

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Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration … [View More]because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0 [View Less]

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Enum module and parallel branching
by Loic Chabert 17 May '16

17 May '16

Hello, I'm trying to use enum module at core level to route my numbers to end-point asterisk servers. As i read (and validated by tests), when a phone number has two DNS NAPTER entries, enum module create two branch, one for each entries, and send request to asterisk servers (in the same time). Do you know any tricks or functions to disable parallel branching, and make serial branching instead (following responses returned by asterisk) ? I have also change preference and order in dns config … [View More]

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sr-users May 2016