sr-users January 2017

sr-users@lists.kamailio.org

87 participants
111 discussions

mhomed opened sockets
by Diego Nadares 30 Jan '17

30 Jan '17

Hi Guys, We have two network interfaces. One public and one private. We had enabled mhomed and works great BUT if we do a netstat -naup | grep 5060 we see that some sockets keeps opened also when the call is finished. The source port in netstat it's not 5060 so I suppose that kamailio use another one for something else. Is this the right behavior? How long does this sockets remain opened? Are we missing something in our cfg? 6 2017-01-25 15:52:04.986651 172.16.213.38 172.16.200.159 SIP/SDP 146 Request: INVITE sip:xxx@172.16.200.159:5060 34 2017-01-25 15:52:19.636895 172.16.213.38 172.16.200.159 SIP 716 Request: ACK sip:xxx@172.16.200.159:5060 | 37 2017-01-25 15:54:38.947831 172.16.208.111 172.16.213.38 SIP 736 Request: BYE sip:127.0.0.8:5060;line=sr-987jhlk* | 40 2017-01-25 15:54:38.950899 172.16.213.38 172.16.200.159 SIP 660 Request: BYE sip:172.16.208.111:5060 | [root@dwrfsd01 kamailio]# netstat -naup | grep 5060 udp 0 0 172.16.213.38:*56086* <http://172.16.213.38:56086> 172.16.200.159:5060 ESTABLISHED 17015/kamailio udp 0 0 172.16.213.38:5060 0.0.0.0:* 17006/kamailio udp 0 0 172.16.213.38:*41971* <http://172.16.213.38:41971> 172.16.200.159:5060 ESTABLISHED 17051/kamailio udp 0 0 172.16.213.38:*54651* <http://172.16.213.38:54651> 172.16.200.159:5060 ESTABLISHED 17008/kamailio Thanks in advance! Diego.

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Presence Module do not load db_url
by Vu. Minh Cao 29 Jan '17

29 Jan '17

Hi, I got error when loading presense module in Kamailio 4.4: [Configure] loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:kamailiorw@127.0.0.1/kamailio") [Error] ERROR: presence [presence.c:608]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) Which is really strange since according to the source code this error just occurs if “db_url” is not set. I already checked my database also, I can access mysql with about username/password. Yours sincerely, — Vu

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rtpengine - Key file error
by Serhat Guler 29 Jan '17

29 Jan '17

Hello, I am trying to setup my Kamailio environment on a new Debian system. Everything went well, except I started facing a problem with the rtpengine. It did install it fine and I can view the program parameters via the help menu; however, when I run it with this command "rtpengine --interface 192.168.0.66 --listen-ng=127.0.0.1:22222 -m 30000 -M 35000 -L 2" or any other commands, it gives an error as "CRIT: Fatal error: Bad command line: Key file does not have group 'rtpengine'". Where is this key file it is complaining about ? I couldn't find anything neither in the doc nor anywhere else. Cheers, Serhat

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missing memory after tls cleanup
by Andy 28 Jan '17

28 Jan '17

Hi, the tls total memory before and after a tls connection is cleaned up does not match, during test there were 10 tls connection created for which I see tls_h_close and tls_h_tcpconn_clean were called exactly 10 times for the same connection ids, bit before and after the tls shm memory does not match, using command: kamcmd mod.stats tls shm could there be a leak? please suggest thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/missing-memory-after-tls-cleanup-tp… Sent from the Users mailing list archive at Nabble.com.

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Ansible Kamailio role
by Alberto Llamas 27 Jan '17

27 Jan '17

Hello Kamailians, I just wanted to share an Ansible role for the installation and compilation of a Kamailio server. It may required improvements and I will be more than happy to heard them. *https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/ <https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/>* Cheers, -- Alberto Llamas Telecommunications Engineer dCAA|dCAP|KPAC|SSCA

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Re: [SR-Users] FOSDEM 2017
by DanB 26 Jan '17

26 Jan '17

Hey Daniel, Would be there 2 persons for sure, so please count us also in. Thanks, DanB

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parallel media forking during ringing phase
by Tomas Zanet 26 Jan '17

26 Jan '17

Hi all, I'm writing to the community because I need a hint from Kamailio and voip gurus: the goal is adding parallel forking call with early-media to our PABX where there's a Kamailio instance running. I mean media streams should be sent to all call destinations before answering the call order, for instance, to see the incoming video. As far as I know, from signaling side should not be a problem because, using Kamailio, there’s append_branch function which forks many branches to all users. The problem is related to media streams because there’s no way to fork media to all users after 180 + SDP or 183, right? Obviously all users must negotiate the same media because media transcoding is basically impossible, especially for video streams. I think there are two ways to do that: 1) using Kamailio and Asterisk. Asterisk has Queue function with ringall strategy, but I’m not sure if Asterisk supports multi domain scenarios. Have ever used Asterisk with Kamailio in multi domain scenarios? 2) enhancing rttpproxy-ng. I don't really know the effort to achieve that. In your opinion, what’s the easiest way to achieve that? Thanks, Best Regards. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

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How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image001.jpg@01D27631.42773770] From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Ryan Wagoner Sent: Tuesday, January 24, 2017 8:26 AM To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Asterisk Proxy Multiple Devices / BLF Issues I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { return; } } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Thanks, Ryan

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kamctl cr reload got 500 failed to re-built tree
by Igor Potjevlesch 25 Jan '17

25 Jan '17

Hello, With more than 173k entriers into carrierroute, now I'm doing "kamctl cr reload", I got the following reply: 500 failed to re-built tree, see log What could going wrong? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

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Issue with From address being modified in ACK when UAC module used version 4.1
by Jonathan Hunter 25 Jan '17

25 Jan '17

Hi Guys, running kamailio 4.1.4 and using uac_replace_from, I am seeing a strange issue with the proxying of an ACK message back from a carrier to freeswitch on the ingress path into a network. So its just a normal call inbound, where on outbound leg we modify the From address, on the inbound leg all remains the same. Now after the ingress side receives the 200ok, it sends an ACK as below; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+44792498881474@192.168.24.8:5080;user=phone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Route: <sip:192.168.24.8;lr=on;ftag=carrier.peering.telecom.im+3+863d20a3+1b9801d2;vsf=AAAAAAAAAAYNBgAPAAgLAgZ3UTMACgBXFUsVFwwcDkAKTwMZGh0YAA8KDkEEQ1NYC0NDXgdOFRpSHg1vbmU-> Content-Length: 0 Max-Forwards: 68 However kamailio changes the From address; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 109.73.69.165:5060;branch=z9hG4bKc1ce.47974fc3da2b669a78f2dcc9a057a127.0 Via: SIP/2.0/UDP 4.4.4.4:5060;rport=5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+441444680332@es132y$}-9>.8n?~9,*%(;zyk393;7e&C^NRone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Content-Length: 0 Max-Forwards: 67 Causing FreeSWITCH to not recognise the request, and therefore not send an ACK. There are no rules set against the ACK processing. Has anyone seen this before? We dont know when it started happening which doesnt help, I will look to setup debug on test environment but just wondered if this is an issue thats been seen before? Many thanks in advance. Jon

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sr-users January 2017