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37510 discussions

Re: [SR-Users] Regarding Opensource DID Routing Solution using Kamailio
by surendra tiwari 06 Nov '17

06 Nov '17

Hi All, I have used the kamailio and redis to create the DID routing solution.please find below in github.this repository is created to help the person who want to setup there own DID routing solution with following feature: 1. Inbound Termination with Carrier IP validation. 2. Carrier LCR for DID/TFN to PSTN forwarding. 3. Inbound Abuse Block. 4. CDR in MongoDB. 5. IPTables Block for Sip-Scanners. 6. Integration with RtpEngine. 7. RedisDB for quick DB Access. https://github.com/surendratiwari3/KamInboundSIP On Mon, Nov 6, 2017 at 9:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > your email was addressed to kamailio-announce mailing list (I took your > message from its bounces filter), but that list is only for announcement > issued by Kamailio project (major news about our releases, security > incidents, events, etc...). > > News from community members are welcome and can be sent to: > > - sr-users(a)lists.kamailio.org for non-commercial announcements > > - business(a)lists.kamailio.org for announcements related to commercial > offerings/products (but here you can also send non-commercial announcements) > > Your announcement is not commercial and can be sent to sr-users. Once it > will be there, I will also make a news article on kamailio.org website. > > I didn't have the time yet to check in detail what you put together, but > sounds interesting. > > Thanks for sharing, > Daniel > > (cc-ed Fred as he is moderator for kamailio announce mailing list) > > On 01.11.17 18:46, surendra tiwari wrote: > > Hi All, > > I have used the kamailio and redis to create the DID routing > solution.please find below in github.this repository is created to help the > person who want to setup there own DID routing solution with following > feature: > > > 1. Inbound Termination with Carrier IP validation. > 2. Carrier LCR for DID/TFN to PSTN forwarding. > 3. Inbound Abuse Block. > 4. CDR in MongoDB. > 5. IPTables Block for Sip-Scanners. > 6. Integration with RtpEngine. > 7. RedisDB for quick DB Access. > > > https://github.com/surendratiwari3/KamInboundSIP > > -- > With Regards, > Surendra Tiwari > VoIP Application Developer, > +91-9967609476 <+91%2099676%2009476> > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > -- With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

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Sipcapture HEP Values
by Soltanici Ilie 06 Nov '17

06 Nov '17

Hi all, How can i get the Destination IP from HEP packet? I tried $hep(dst _ip) like in documentation (https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#sipcapture_mo…) - but is not working. (pvar "hep" has an invalid name param [dst_ip]). Thank You.

1 0

Invalid JSON when dumping htable with JSONRPC-S
by Marrold 03 Nov '17

03 Nov '17

Hi All, I am currently trying to retrieve the contents of a htable using the JSONRPC-S module over HTTP, however I have observed if a slot contains multiple values, the JSON returned has duplicate keys which could be considered invalid. Here are the relevant lines from kamailio.cfg - event_route[xhttp:request] { if ($hu =~ "^/carrier") { jsonrpc_exec('{"jsonrpc": "2.0", "method": "htable.dump", "params" : {"htable": "carrier"}, "id": 1}'); xhttp_reply("200", "OK", "text/html", "$jsonrpl(body)"); } } And the output, showing a slot with two items- { "entry": 5, "size": 2, "slot": { "item": { "name": "carrierA", "value": "90", "type": "str" }, "item": { "name": "carrierB", "value": "190", "type": "str" } } } And the same from `kamcmd -s unix:/tmp/kamailio_ctl htable.dump carrier` - { entry: 5 size: 2 slot: { item: { name: carrierA value: 90 type: str } item: { name: carrierB value: 190 type: str } } } Would it be possible to change it to something like the following - { "entry": 5, "size": 2, "slot": [ { "name": "carrierA", "value": "90", "type": "str" }, { "name": "carrierB", "value": "190", "type": "str" } ] } Any other suggestions or work arounds would be appreciated, Thanks Matthew

2 1

Contacts in 302
by Diego Nadares 03 Nov '17

03 Nov '17

Hi Guys, I'm generating a 302 reply from kamailio. In this 302 I append new branches with new contacts. if($var(routing)=~"redirect"){ jansson_get("contacts_len", "$var(evmsg)", "$var(contacts_len)"); xlog("L_INFO", "Contacts len $var(contacts_len)"); $var(i) = 0; while ($var(i) < $var(contacts_len)){ jansson_get("contacts[$var(i)]", "$var(evmsg)", "$var(contact)"); append_branch($var(contact), "0.5"); $var(i) = $var(i) + 1; } send_reply("302", "Moved Temporarily"); exit; } The problem i'm facing is that I can't delete the original contact *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>* The resulting conctact: Contact: *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>*, <sip:1111@188.111.111.112>;q=0.5, <sip:1111@188.111.111.112>;q=0.5 I all ready tried with remove_hf('Contact') and adding the new ones after that but it doesn't worke either. The problem is that one gw takes the first contact over and over again and never the other two. Thanks in advance. Diego.

2 4

Multiple IPs
by Oz Mortimer 03 Nov '17

03 Nov '17

Hi all, I have Kamailio set as an SBC with freeswitch behind - that all works fine. Freeswitch can send calls out VIA Kamailio to external IP address’, but if I try sending a call to another IP hosted by Kamailio the seems to go nowhere… Supplier -> Kamailio -> Customer1 -> Kamailio -> Customer2 The Kamailio server has multiple assigned IPs, which are used to multi-tennant - but I cant for the like of me to get Tennant to tenant calls to work (via freeswitch) Any pointers?

2 13

onreply_route block not processing replies
by José Seabra 03 Nov '17

03 Nov '17

Hi there, I'm facing an issue regarding with replies coming to kamailio that aren't processes inside of onreply_route block. Anyone here can help me understand why these replies (1XX) aren't entering on onreply_route bock? is there any situation already identified on Kamailio that can originate this behavior? The Kamailio version used is 5.0.1. Thank you Regards -- José Seabra

2 3

Branching Kamailio v5.1
by Daniel-Constantin Mierla 03 Nov '17

03 Nov '17

Hello, I am planning to create the got branch 5.1 (to be used for v5.1.x series) by end of next week (likely on Friday, Nov 10, 2017). After that, the master will be open for new features and fixes will need to be backported to 5.1. Should anyone want a different schedule, reply to the mailing lists and propose your plans. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WebRTC error code 488 in iOS application
by Denys Pozniak 03 Nov '17

03 Nov '17

Hello! Probably this case is not connected Kamailio directly, but it can be somebody will point me in the correct direction :) We have VoIP solution: FreeSWITCH -> Kamailio + Rtpengine -> WebRTC Chrome. Everything works Ok except iOS application based on React Native WebRTC ( https://github.com/oney/react-native-webrtc) + JsSIP. It answers error 488 on incoming INVITE (wSDP): INVITE SDP to iOS: *......* *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* iOS console log: *11:55:23.670 browser.js:133 JsSIP:WebSocketInterface send() +10ms* *11:55:23.670 browser.js:133 JsSIP:RTCSession session progress +5ms* *11:55:23.671 browser.js:133 JsSIP:RTCSession emit "progress" +1ms* *11:55:23.671 webrtc.js:108 WebRTC.on progress: local* *11:55:23.679 browser.js:133 JsSIP:RTCSession emit "peerconnection" +8ms* *11:55:23.680 webrtc.js:89 WebRTC.on peerconnection* *11:55:23.680 browser.js:133 JsSIP:RTCSession emit "sdp" +1ms* *11:55:23.689 browser.js:133 JsSIP:Transport send() +20ms**11:55:23.689 browser.js:133 JsSIP:Transport sending message:* *SIP/2.0 488 Not Acceptable Here* *11:55:23.701 debug.js:115 JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:Error: Failed to set remote offer sdp: Failed to create channels.* I found similar issue on thier git https://github.com/oney/react-native-webrtc/issues/293 and main solution is to struct SDP in way like: *a=group:BUNDLE audiom=audio PORT RTP/SAVPF N M Ka=mid:audio* How to modify SDP on Kamailio side (after Rtpengine modification) like below? *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *a=group:BUNDLE audio* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=mid:audio* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* BR, Denys

1 1

User Registration with location awareness
by Daisy Stevens 03 Nov '17

03 Nov '17

Hi Guys Please forgive me if I seem to asking stupid questions but I am a complete dummy at the moment regarding this software. I would love to be able to set up two kamailio servers with the idea of expansion in the future to handle SIP registrations, however having a two server model how would I get the calls to route to another user that is registered on the other kamailio box? Kind Regards Daisy -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

2 1

msilo Content-Type for picture and video?
by sscc 03 Nov '17

03 Nov '17

Hello kamailio folks i am successfully storing and dumping messages for off line users using msilo content type text/plain. i want to store off-line messages for pictures and videos. my particular queries are 1) is msilo store picture and videos? (i have observed blob in silo database table which is used for pictures. 2) in case it does what will be declared in content type hdr parameter? any help in this regard will highly be appreciated. regards, sscc -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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