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37547 discussions

rtpengine upgrade fails
by Juha Heinanen 16 Jul '14

16 Jul '14

i build debian packages for rtpengine branch 3.3.0.0+0~mr3.4.1.0. after that i did 'apt-get dist-upgrade' and it failed like this: Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Installing new version of config file /etc/init.d/ngcp-rtpengine-daemon ... Restarting RTP/media proxy: ngcp-rtpengine-daemonFATAL: Module xt_MEDIAPROXY not found. iptables: No chain/target/match by that name. ip6tables: No chain/target/match by that name. FAILED TO CREATE KERNEL TABLE 0, KERNEL FORWARDING DISABLED invoke-rc.d: initscript ngcp-rtpengine-daemon, action "restart" failed. dpkg: error processing ngcp-rtpengine-daemon (--configure): subprocess installed post-installation script returned error exit status 255 however, if i now run 'apt-get install ngcp-rtpengine-daemon', setting up ngcp-rtpengine-daemon succeeds and also packages that depended on it get installed fine: Reading package lists... Done Building dependency tree Reading state information... Done ngcp-rtpengine-daemon is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. 2 not fully installed or removed. After this operation, 0 B of additional disk space will be used. Do you want to continue [Y/n]? Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Restarting RTP/media proxy: ngcp-rtpengine-daemon. Starting RTP/media proxy: ngcp-rtpengine-daemon already running. Setting up ngcp-rtpengine (3.3.0.0+0~mr3.4.1.0) ... any ideas why setting up ngcp-rtpengine-daemon fails with dist-upgrade? -- juha

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KAMILIO-4.1.4 REGISTRATION ISSUE
by Cibin Paul 16 Jul '14

16 Jul '14

I have kamailio 4.1.4 installed with DB Engine as mysql. I added users using the command kamctl add and I could see the entry in subscriber table. When I am trying to register, kamailio is accepting any combination of username and password not listed in the table. What could be wrong? I followed the INSTALLATION document provided in the source file. Please help me troubleshoot this. Kamailio is up and running and not throwing any errors. kamctl moni shows 1 user registered. But the registered user doesn’t even exist Server Details OS : Centos 5.5 Kamailio version - 4.1.4 MySQL version - 5.0.95 Cibin

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Rtpproxy redundancy
by Dren Daka 16 Jul '14

16 Jul '14

Hi, I'm using kamailio with two network interfaces also running rtpproxy on the same interfaces so rtpproxy is load balancing, when one of the connections is down rtprpoxy still sends 50% of the calls in that interface. How can i make rtprpoxy to do failover from the non working connection to the working one instead of load balancing. Thanks.

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BUG: ip_addr2sbuf: unknown address family 0
by AliReza Khoshgoftar Monfared 16 Jul '14

16 Jul '14

Hi, I am running a simple command on timer in a kamailio proxy. Here is the related parts of the config scripts: loadmodule "timer" > modparam("timer", "declare_timer", "tmr1=LOG_STATS,1,fast,enable"); > > route{ > ... > } > > route[LOG_STATS] { > exec_avp("netstat -ul | grep ':sip' | awk '{print > $$2}'","$avp(s:test)"); > xlog("L_ALERT","logged RecvQ as $avp(s:test)"); > return; > } > Each time I run the "netstat" using the exec() inside the timer route block, I get the following error from Kamailio in my log: kamailio[28546]: : exec [../../parser/../ip_addr.h:669]: ip_addr2sbuf(): > BUG: ip_addr2sbuf: unknown address family 0 > What is the meaning of this error, and what does kamailio mean by reporting it as a "BUG"? Is there any obvious reason, I am getting this? Is it related to calling netstat to survey the RecvQ? Thanks, Alireza

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have problem with iptel account
by Samhu 16 Jul '14

16 Jul '14

hi, This is Sam from Draytek Crop. We have a IPPBX device with iptel for SIP account as below. User B dial to IPPBX could heart the answer from Auto Attendant of IPPBX. Then user B cannot heart anything (after input the extension of user A) when user A pickup his telephone . At this momemt, user A could hear everything from user B. This Network Architecture worked fine before. And it still works fine now if we select the account of "draytel.org". Please show us how to solve the problem. Best Regards, Sam Hu

2 1

SCA and PRESENCE
by Joel White 15 Jul '14

15 Jul '14

I am having an issue where I enable presence modules and the SCA module stops working. I disable presence and SCA works fine on Polycom phones. Does anyone know what I am missing or doing wrong? Thank you in advance Joel

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function to reset all flags or bflags?
by Juha Heinanen 15 Jul '14

15 Jul '14

there is a function to reset a flag or bflag, but i haven't found one that would reset all flags or bflags. is there a way to do it? if not, is it ok to add such function? -- juha

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Many instances of "new URI shorter than old URI" in log
by Alex Villacís Lasso 15 Jul '14

15 Jul '14

Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:49 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI I am not really sure of what this message is supposed to indicate. It looks something related to uac_restore_[from,to] functions, but some debugging statements just before the use show that the message is not originating from there. This message is showing up hundreds of times per day.

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set_rtp_proxy_set behaviour
by Igor Potjevlesch 15 Jul '14

15 Jul '14

Hello, I set two entries of rtpproxy like : modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") For a particular case, I want to use the second RTPProxy and the first one as the default one. If in a conditional block I set set_rtp_proxy("2"), all other call to rtpproxy_manage will use the first RTPProxy? Or there is a load-balancing introduce by default. Regards, Igor.

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Routing calls to Asterisk using dispatch
by Olli Heiskanen 15 Jul '14

15 Jul '14

Hello, I've started playing with an idea to add multiple asterisk servers and using dispatcher to balance the sip load between them. I added the code according to dispatcher module documentation ( http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I think there's something off in my setup: kamctl ul output shows 2 AORs for one client: AOR:: 770(a)testers.com Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP Q= Expires:: 3221 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35ae-2a2 Reg-Id:: 0 Last-Keepalive:: 1405174150 Last-Modified:: 1405174150 AOR:: 770(a)1.1.1.1 Contact:: sip:770@1.1.1.1:5070 Q= Expires:: 68 Callid:: 327fcf07641f80006e962821112a61b5(a)testers.com Cseq:: 754 User-agent:: Asterisk PBX 11.10.2 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35af-a82 Reg-Id:: 0 Last-Keepalive:: 1405174477 Last-Modified:: 1405174477 I don't think I should be seeing an AOR for 770 where Contact is the public address of my server (here 1.1.1.1) and User-Agent which is Asterisk. I'm using Asterisk Realtime integration, and by what I can tell the sip messages are going nicely, client authenticates with Kamailio and sends this message to Asterisk (which is on the same machine; Kamailio at 5060 and Asterisk at 5070): 1.1.1.1.sip > 1.1.1.1.vtsas: SIP, length: 374 REGISTER sip:1.1.1.1:5070 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKbc8a.f4473947000000000000000000000000.0 To: <sip:770@1.1.1.1> From: <sip:770@1.1.1.1>;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1 CSeq: 10 REGISTER Call-ID: 7ffa0191-13742(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 Contact: <sip:770@1.1.1.1:5060> Expires: 3600 Currently I can make calls from 770 to 123 which is an Asterisk extension that answers, plays hello world and hangs up. However I can't call another sip clients when I route calls through Asterisk, they do work fine if I don't use Asterisk for handling calls, but I'd like Kamailio to be in the role of proxy/loadbalancer and Asterisk to handle calls. My config is the simple default config, added with realtime stuff and then dispatcher according to the documentation. I wonder if there's something wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something else going wrong? Has anyone seen results like this and do you spot something here that needs fixing? Thanks, Olli

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