sr-users April 2018

sr-users@lists.kamailio.org

81 participants
104 discussions

rtpengine installation dependencies
by Aqs Younas 05 Jul '18

05 Jul '18

Greetings list. This might not be related to Kamailio but I thought someone would be able to give a hand in resolving dependency during the installation of rtpengine on Debian 8 Jessie. It is complaining about below dependencies. root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage dpkg-buildpackage: source package ngcp-rtpengine dpkg-buildpackage: source version 6.3.0.0+0~mr6.3.0.0 dpkg-buildpackage: source distribution unstable dpkg-buildpackage: source changed by Sipwise Jenkins Builder < jenkins(a)sipwise.com> dpkg-buildpackage: host architecture amd64 dpkg-source --before-build rtpengine dpkg-source: info: using options from rtpengine/debian/source/options: --extend-diff-ignore=.gitreview dpkg-checkbuilddeps: *Unmet build dependencies: debhelper (>= 10~) libbcg729-dev* dpkg-buildpackage: warning: build dependencies/conflicts unsatisfied; aborting dpkg-buildpackage: warning: (Use -d flag to override.) I tried to install debhelper from deb but complained from dh-autoreconf and that itself complained abount debhelpler. So, no one is being installed. Also, If someone can help me how to resolve dependency for libbcg729-dev too. Any pointer or hint or link to some installation guide would be more than welcome. Br, Aqs.

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Re: [SR-Users] ASN.1 HI2- IRI-Parameters
by Mojtaba 06 Jun '18

06 Jun '18

Any idea in this regards? I understood it's structure of HI2Operations as ASN.1 IRI-HI2, I attached it. I think i should convert SIP signalling flow to this structure for transmission. Is it right way that i am doing? With Regards.Mojtaba On Thu, Apr 26, 2018 at 12:50 PM, Mojtaba <mespio(a)gmail.com> wrote: > Hello, > The declaration of details are in 3GPP TS 101-671 Annex D.5 (ASN.1 > description of IRI (HI2 interface)). > I need a sample packet of HI2Operations for underestanding, so that > i'll develop this HI in kamailio. > With Regards.Mojtaba > > > On Thu, Apr 26, 2018 at 11:20 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> can you provide more details or references (web links) about the operations >> you want to do? Like what sip message comes to kamailio and what you need to >> change to it. >> >> Cheers, >> Daniel >> >> >> On 25.04.18 06:00, Mojtaba wrote: >> >> Hello. >> I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose >> anybody have experience in this regards? >> Thanks. >> Mojtaba >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com > > > > -- > --Mojtaba Esfandiari.S -- --Mojtaba Esfandiari.S

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Kamailio Behind TCP Load Balancer
by Colin Morelli 04 Jun '18

04 Jun '18

Hey all, I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware. My application is deployed in AWS and I'm tying to place Kamailio behind an ELB. For the most part, everything is fine. For my specific implementation I'm disabling UDP as a signaling transport and using only TLS. This enables me to not have to worry about a SIP-aware LB at the edge because replies to an incoming request will be sent over the existing established TCP socket (avoiding any crazy routing requirements). However - this poses an issue with source addresses. Does Kamailio support anything like the proxy protocol ( http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP stream information from a load balancer? Do I need to go back to exposing it directly to the world so that I can get source addresses? Thanks, Colin

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Kamailio v5.1.3 Released
by Daniel-Constantin Mierla 07 May '18

07 May '18

Hello, Kamailio SIP Server v5.1.3 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.3. For more details about version 5.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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 Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Determining next hop for any SIP message
by George Diamantopoulos 07 May '18

07 May '18

Hello all, I've been trying to figure out a cleaner way to determine the next hop for a SIP message, mainly for use within the NATMANAGE route for multi-homed kamailio instances (with three or more network interfaces on the kamailio host). So far I have achieved this with a series of nested ifs, depending on whether the message is a request or a response and by calculating the next hop based on the various headers (R-URI, Route) and variables ($T_req($Ri), $dd, ) involved in SIP routing. A simpler way to do it, of course, would be to use the onsend_route, but that would most likely introduce an unnecessary overhead for all routed messages. I recently noticed there a pseudovariable called $nh(key), and I believe I can use $nh(d) to the same effect. I understand however, that this works for requests only. Also, the description of this PV in the documentation reads as follows: Return attributes of next hop for the SIP request. Address is taken from > dst_uri, if set, if not from new r-uri or original r-uri. > What is not clear to me is if this covers in-dialog requests with the Route header set as well. Does the inclusion of a route header set the dst_uri PV? And if yes, is it safe to rely on dst_uri during request processing or is it set only after completion of script processing? Lastly, is there an analogue for SIP responses? If not, is it safe to rely on first Via header to determine next hop for responses, or are there any other corner cases I need to heed? Thanks in advance, BR, George

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How long time does a push take?
by Kjeld Flarup 07 May '18

07 May '18

Hello I have a question to those of You who uses push via Google or Apple to initiate calls. How long time does it usually take, before the phone gets the push and wakes up? Our experience is, that it can take from 2 to 10 seconds, and that not impressing when we are talking about telephony. -- -------------------- 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

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MSILO: Message limit on SIP stored offline MESSAGE delivery
by Abdul Basit 04 May '18

04 May '18

Hi Team, I have setup IM server with kamailio using MSILO module. Its working fine now. I have a scenario where lot of offline messages are being stored in mysql database for a user B. As soon as destination (B) REGISTER with kamailio, these messages will deliver to him. Now, kamailio will deliver all of these messages that will cause user (B) device (softphone on mobile) stuck or freezs for the moment messages are being delivered like bombardment from kamailio. Is there any way, where I can define batch size of messages to be delivered after provided interval? like batch_size=10 messages and batch_interval= 5sec, etc. I want to avoid applying static limit to mysql query in module source code. Please advise. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

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DMQ and/or Database for dialogs, registrations, etc..
by Joel Serrano 03 May '18

03 May '18

Hi all, Just wanted to know what your opinions were on using DMQ modules over database for things like dialog replication, registrations, etc... Is DMQ the "new way to go"? I know that there lots of ways of doing things with each having pros/cons... But I was wondering... What does the community think on this topic? Are you guys taking advantage of the DMQ modules or are you still relying on database as much as possible? Maybe a combination of both? Cheers, Joel.

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BLF+Asterisk+NAT
by John Bob 03 May '18

03 May '18

Phone sends Subscribe Kamailio Responds 202 OK Kamailio Responds with NOTIFY Via: SIP/2.0/TCP 66.66.66.66:5051;branch=z9hG4bKfe3e.e4fedb84000000000000000000000000.0 To: <sip:30041@66.66.66.66 <mailto:30041@66.66.66.66>>;tag=48ae253ab7 From: <sip:47701@66.66.66.66 <mailto:47701@66.66.66.66>>;tag=00c39aabf419ddeab4250cd65af8ef90-6fb2 CSeq: 2 NOTIFY Call-ID: 1493ef52c41b6c1a Content-Length: 0 User-Agent: kamailio (5.1.2 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:192.168.1.30;transport=udp> Subscription-State: terminated;reason=timeout In that packet I see Subscription-state: terminated;reason=timeout Im not sure what I am missing or misconfigured. Any help would be great. I’ve attached my config. Thanks.

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Send UPDATE to inform changes in SDP
by Alfonso Pinto 30 Apr '18

30 Apr '18

Hi guys, I'm trying to setup a Call Forward No Reply with Kamailio and I've run into this problem (that was expected): When A INVITEs B, B replies with 183 and some SDP. B doesn't answer so Kamailio triggers the timeout and sends the INVITE TO C. C replies with 200 OK and some SDP, but A side drops the call as the media session doesn't match with the expected one. To solve it, I'm trying to send to A an UPDATE before relaying the 200 OK from C. I tried to use UAC module (uac_req_send + $uac_req) but I don't see how I can UPDATE the existing dialog. uac_req_send seems to try to setup a new dialog always. Is there a way to indicate to uac_req_send that I wan't to use the existing dialog? Or maybe there is another mechanism I can use to send the UPDATE inside the dialog? Regards, Alfonso.

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sr-users April 2018