Re: [SR-Users] Help with KEEP-ALIVE method

Thank you for clarification. -- Best regards, Sergey Basov e-mail: sergey.v.ba...@gmail.com 2017-03-01 20:05 GMT+02:00 Victor Seva : > 2017-03-01 15:48 GMT+01:00 Sergey Basov : >> 2017-03-01 15:57 GMT+02:00

Re: [SR-Users] Help with KEEP-ALIVE method

2017-03-01 15:48 GMT+01:00 Sergey Basov : > 2017-03-01 15:57 GMT+02:00 Daniel-Constantin Mierla : >> If yes, this is not a valid SIP message, because it lacks mandatory >> headers such as call-id, cseq, from/to. >> > Yes it is without any headers... So

Re: [SR-Users] Help with KEEP-ALIVE method

Hi, Daniel Yes it is without any headers... I have attached screenshot from wireshark, I can not save it because this is sip tls... Thank you -- Best regards, Sergey Basov e-mail: sergey.v.ba...@gmail.com 2017-03-01 15:57 GMT+02:00 Daniel-Constantin Mierla

Re: [SR-Users] Help with KEEP-ALIVE method

Hello, On 28/02/2017 17:05, Sergey Basov wrote: > Hi All. > > Today I have problem with connection from 1 of the clients. > Their PBX sends KEEP-ALIVE after some time after REGISTER. > > I have next error in kamailio log > > Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR: >

[SR-Users] Help with KEEP-ALIVE method

Hi All. Today I have problem with connection from 1 of the clients. Their PBX sends KEEP-ALIVE after some time after REGISTER. I have next error in kamailio log Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR: [tcp_read.c:1354]: tcp_read_req(): bad request, state=7, error=4

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

January 2017 04:27 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers Hi, just remove: #!define WITH_ASTERISK From your kamailio.cfg and restart it. -- Daniel Grotti On 01/02/2017 06:36 PM, Manoj Gupta wrote

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

sage. -Original Message- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 02 January 2017 10:52 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers Kamailio-asterisk

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

0:52 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers Kamailio-asterisk doc: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb There are tones of documentation about kamailio out there. Cons

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Daniel Grotti Sent: 02 January 2017 10:34 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers You should add "ims.airtel.in" as kamailio local domain, in your kamailio.domain table. -- Daniel Grotti On

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

er.org] On Behalf Of Daniel Grotti Sent: 02 January 2017 10:34 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers You should add "ims.airtel.in" as kamailio local domain, in your kamailio.domain table.

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

all copies of the original message. -Original Message- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 02 January 2017 08:54 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VO

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

nuary 2017 08:54 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers Hi, please configure this in your kamailio.cfg: debug=3 # debug level, 1 is low and 4 is high (lots of output) log_facility=LOG_LOCAL7 ... l

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

el Grotti Sent: 02 January 2017 08:54 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers Hi, please configure this in your kamailio.cfg: debug=3 # debug level, 1 is low and 4 is high (lots of output) log_facility=L

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

al Message- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 02 January 2017 08:10 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers Hi, have you configured kamail

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

=5 memlog=5 #log_facility=LOG_LOCAL0 log_facility=LOG_LOCAL6 Manoj K. Gupta -Original Message- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 02 January 2017 08:10 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help Asterisk

[SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1:

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Hi, have you configured kamailio in order to log to /var/log/kamailio instead of syslog ? https://www.kamailio.org/dokuwiki/doku.php/utils:basic-syslog-configuration -- Daniel Grotti On 01/02/2017 03:36 PM, Manoj Gupta wrote: Request to all – Please help we are BADLY stuck in this

[SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1:

[SR-Users] help configure dispatcher in Amazon ec2

Hi, I’m very appreciated if someone can figure out this problem for me. My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system like this: soft phone 1(Linphone) —> | Load balancer—> | Sip Server huancomputer

Re: [SR-Users] help with string modification

Hi, All. One more question related to remove_hf... I have added route: # Fix user-agent and server route[RemoveHeader] { remove_hf("server"); remove_hf("user-agent"); return; } I use it form request_route { route(RemoveHeader); . } failure_route[--- all what i

Re: [SR-Users] help with string modification

Thank you Daniel. Is it safe to use remove_hf("User-Agent") without check if this header exist? or better use if(is_present_hf("User-Agent")) { remove_hf("User-Agent"); } ? Thank you. 25 нояб. 2016 г. 2:56 PM пользователь "Daniel Tryba" написал: > On Fri, Nov 25, 2016 at

Re: [SR-Users] help with string modification

On Fri, Nov 25, 2016 at 02:08:07PM +0200, Sergey Basov wrote: > Hello All. > > I have some troubles with upstream sip switch. > It ignores SIP packets which contains: > > User-Agent: FPBX-2.11.0(11.17.1) > or > Server: User-Agent: FPBX-2.11.0(11.17.1) > > If space is present before first "("

[SR-Users] help with string modification

Hello All. I have some troubles with upstream sip switch. It ignores SIP packets which contains: User-Agent: FPBX-2.11.0(11.17.1) or Server: User-Agent: FPBX-2.11.0(11.17.1) If space is present before first "(" then sip switch works as expected So my question is: how corektly make analyze and

Re: [SR-Users] help with NOTIFY

you. volga629 From: "volga629" <volga...@skillsearch.ca> To: "sr-users" <sr-users@lists.sip-router.org> Sent: Wednesday, 19 October, 2016 18:36:52 Subject: [SR-Users] help with NOTIFY Hello Everyone, I am trying forward NOTIFY to client. My setup

[SR-Users] help with NOTIFY

Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server local lan ---kamailio internet client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 ->

Re: [SR-Users] help with kamailio rpm made from source

Hello I tested 4.3 and 4.4 and kamailio -E -DDD gives the sameI compiled from source - the same results:[root@kazootest2 kamailio]# kamailio -E -DDDloading modules under config path: /usr/local/lib64/kamailio/modules/ 0(1) INFO: [sctp_core.c:75]: sctp_core_check_support(): SCTP API not

Re: [SR-Users] help with kamailio rpm made from source

Hello I tested a package http://download.opensuse.org/repositories/home:/kamailio:/v4.3.x-rpms/CentOS_6/x86_64/kamailio-4.3.6-1.1.x86_64.rpm  ( I downlowaded 4.3.6 version rpm and installed it. Kamailio restarts well. The behaviour is the same. the phone registeres without nonce. I install

Re: [SR-Users] help with kamailio rpm made from source

RPMBUILD produces several kamailio rpms   Now I install the following rpms: [root@kazootest3 ~]# rpm -qa | grep kamakamailio-presence-4.3.4-0.x86_64kamailio-4.3.4-0.x86_64kamailio-utils-4.3.4-0.x86_64kamailio-outbound-4.3.4-0.x86_64kamailio-tls-4.3.4-0.x86_64kamailio-kazoo-4.3.4-0.x86_64 still

Re: [SR-Users] help with kamailio rpm made from source

I made loadmodule and modparam("debugger", "cfgtrace", 1) but anyway - no logs when I register. As I understand - it's like no config file. On Thursday, September 15, 2016 6:01 PM, Daniel-Constantin Mierla wrote: I am not familiar with kazoo configs, maybe asking

Re: [SR-Users] help with kamailio rpm made from source

I am not familiar with kazoo configs, maybe asking on their mailing list can help you more. >From Kamailio point of view, you can load debugger module and set its cfgtrace parameter to 1, then see what actions from config are executed and why is not getting to the authentication part. Cheers,

Re: [SR-Users] help with kamailio rpm made from source

here are my "define_with flags" from SPEC file (opensuse one) # list of flags to enable extra packages%define _with_bdb 0%define _with_carrierroute 0%define _with_cnxcc 0%define _with_dnssec 0%define _with_erlang 0%define _with_ev 0%define _with_geoip 0%define _with_java 0%define _with_json

Re: [SR-Users] help with kamailio rpm made from source

/etc/kazoo/kamailio/default.cfg - which containes all routes.2600hz/kazoo-configs | | | | || | | | || 2600hz/kazoo-configs kazoo-configs - Kazoo Configuration Files for Software We Use | | | | I test on a working server (testing one) and a working

Re: [SR-Users] help with kamailio rpm made from source

Hello I took this spec from suse. It generates no errors. When I installed from the RPM I had made - the phone register, but The phone sends a REGISTER and the KAmailio sends 200ok back to the phone (so no NONCE authorization) and no logs during it. In default.cfg I set L_DBG but no logs are

Re: [SR-Users] help with kamailio rpm made from source

Then you just need add those files in various packages inside the spec file, so they are not detected to be orphaned. Maybe you can inspire from: - https://build.opensuse.org/package/view_file/home:kamailio:v4.3.x-rpms/kamailio43/kamailio.spec?expand=1 Cheers, Daniel On 14/09/16 16:35,

Re: [SR-Users] help with kamailio rpm made from source

4.3.4 version is for KazooIt is on production server currently. I need to rebuild the current RPM so as to apply patches. But first I want to get a working Kamailio and only after it I will apply the patches. I think I may take a list of modules from the production Kazoo-kamailio and rearchive

Re: [SR-Users] help with kamailio rpm made from source

Hello, any reason not to use series 4.4.x? Iirc, the latest spec that got update on 4.4 are those for oracle enterprise linux, perhaps is something that you can reuse a lot for upgrading to the centos flavour. On the other hand, you can use opensuse build service if you want to build yourself,

Re: [SR-Users] help with kamailio rpm made from source

Hellowhich SPEC file is used by the Kamailio group to build rpm? On Tuesday, September 13, 2016 7:56 PM, Dmitry wrote: I use Centos 6.7 On Tuesday, September 13, 2016 7:51 PM, Dmitry wrote: Hello I used:

Re: [SR-Users] help with kamailio rpm made from source

I use Centos 6.7 On Tuesday, September 13, 2016 7:51 PM, Dmitry wrote: Hello I used: kamailio-4.3.4_src.tar.gz /kamailio-4.3.4/pkg/kamailio/centos/6/ I found several spec files:[root@kazootest2 kamailio-4.3.4]# find . -name

Re: [SR-Users] help with kamailio rpm made from source

Hello I used: kamailio-4.3.4_src.tar.gz /kamailio-4.3.4/pkg/kamailio/centos/6/ I found several spec files:[root@kazootest2 kamailio-4.3.4]# find . -name

Re: [SR-Users] help with kamailio rpm made from source

Hello, which rpm spec did you use? There are several of them in the source tree, some not really maintained. Cheers, Daniel On 13/09/16 14:33, Dmitry wrote: > Hello, All > > When I take a SPEC file from kamailiotar.gz - during rpmbuild I > encounter: > > Checking for unpackaged file(s):

Re: [SR-Users] help with kamailio rpm made from source

Hello, All When I take a SPEC file from kamailiotar.gz - during rpmbuild I encounter: Checking for unpackaged file(s): /usr/lib/rpm/check-files /root/rpmbuild/BUILDROOT/kamailio-4.3.4-0.0.el6.x86_64error: Installed (but unpackaged) file(s) found:   /usr/lib64/kamailio/modules/auth_xkeys.so   

Re: [SR-Users] help with kamailio rpm made from source

 I see : ERROR: [tcp_main.c:2790]: tcp_init(): bind(9, 0x7fd50bd8ee34, 16) on 127.0.0.1:5060 : Address already in use But I commented out all TCP (listen TCP) so why is this error happen? On Friday, September 9, 2016 10:52 AM, ycaner wrote: Hello; it is

Re: [SR-Users] help with kamailio rpm made from source

Hello; it is clear that kamailio crashs. Could you start with "kamailio -E -ddd" and then see logs. it gives hit. Probably libraries has some conflicts. -- View this message in context: http://sip-router.1086192.n5.nabble.com/help-with-kamailio-rpm-made-from-source-tp151601p151626.html Sent

[SR-Users] help with kamailio rpm made from source

Hello I obtained the source from  https://www.kamailio.org/pub/kamailio/4.3.4/src/kamailio-4.3.4_src.tar.gz I found a SPEC file there and I prepared a RPM with this spec file. When I start a Kamailio (from this rpm) - it starts but the LOG file gives only 1 line:  INFO: [tcp_main.c:4657]:

[SR-Users] Help with dispatcher ping: Kamailo 4.1

Hi, I'm currently running Kamailio 4.1 and was wondering if there is a way to control ds_ping_interval on a dispatcher setid. Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) ac...@thinkingphones.com ___ SIP Express Router (SER) and

Re: [SR-Users] Help with routing block and Dispatcher Module

Thank you very much, I did it this way, and it worked perfect!!! if (is_subscriber("$ru", "subscriber", "2")) { ... logic ... } 2016-06-13 3:05 GMT-06:00 Daniel-Constantin Mierla : > Hello, > > you have to show the request_route block with the part where the >

Re: [SR-Users] Help with routing block and Dispatcher Module

Hello, you have to show the request_route block with the part where the route(DISPATCH) is executed. You can use is_subscriber() to see if the target number is a local subscriber and route via location. Cheers, Daniel On 10/06/16 03:39, pablo rosales wrote: > Hi everyone! I am a newbie with

[SR-Users] Help with routing block and Dispatcher Module

Hi everyone! I am a newbie with Kamailio. I'm trying to connect a kamailio server to an asterisk gateway. We are using basic Kamailio 4.4.x configuration file with mysql and rtpproxy we are having same issues with dispatcher module. we can not call between local subscribers. Basically, If I

Re: [SR-Users] Help with Kamailio app_java module

Hello, can you check if kamailio is actually running? ps auxw | grep kamailioo Also, look inside the syslog file to see what messages are written there by kamailio. Cheers, Daniel On 14/04/16 20:03, Clarence Sandjon wrote: > Hi! > > I am trying to use the app_java module with kamailio-4.4.0

[SR-Users] Help with Kamailio app_java module

Hi! I am trying to use the app_java module with kamailio-4.4.0 SIP server. Every time I load the app_java module, my users can't register and the sever replies with destintion unreachable (port unreachable). How can I fix this issue? Also, how can I use the Kamailio.java file since it doesn't

[SR-Users] Help with Kamailio app_java module

Hi! I am trying to use the app_java module with kamailio-4.4.0 SIP server. Every time I load the app_java module, my users can't register and the sever replies with destintion unreachable (port unreachable). How can I fix this issue? Also, how can I use the Kamailio.java file since it doesn't

Re: [SR-Users] Help with Kamailio java module

I am not using the app_java module, so not familiar with how was built. log_stderr is a variable inside kamailio. Can you share the java code you try to run? Cheers, Daniel On 11/04/16 10:53, Bonjour Madame wrote: > > Thanks for responding. I followed the steps in the readme documents > and

Re: [SR-Users] Help with Kamailio java module

Thanks for responding. I followed the steps in the readme documents and was able to compile my application. However, when I try to run it, I get the error unsatisfiedlinkerror undefined symbol: log_stderr. I don't know what is the reason and how I can solve it. On Apr 11, 2016 1:05 AM,

Re: [SR-Users] Help with Kamailio java module

Hello, I see the module has some txt docs in the folder: https://github.com/kamailio/kamailio/tree/master/modules/app_java Is it what you have tried? Cheers, Daniel On 09/04/16 08:10, Bonjour Madame wrote: > > Hi! > > > > I am a newbie and I need help understanding how to use the java

[SR-Users] Help with Kamailio java module

Hi! I am a newbie and I need help understanding how to use the java module with Kamailio. I am trying to create a basic java application that will print the SIP message type being processed by Kamailio. I read the app_java module but can't still figure it out. Any help or reference will be

Re: [SR-Users] Help

[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of SamyGo Sent: Sunday 28 February 2016 15:51 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users@lists.sip-router.org> Subject: Re: [SR-Users] Help Hi, I think the best guide closest t

Re: [SR-Users] Help

b 2016 15:35:50 -0500 Subject: [SR-Users] Help Hi, I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some ki

Re: [SR-Users] Help

ject: [SR-Users] Help Hi, I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of S

[SR-Users] Help Required

Hi All. Does anyone know where to find training guide for Kamailio HA? I want to have 2 Kamailio in HA with four (4) Freeswitch servers behind them. I will also like to customer cgrate to manage the accounting side in a multi-tenant environment. I am just researching for this on my own and I

Re: [SR-Users] Help with sip balancer

Thankyou Alexandru for your suggestions. I'll give it a try tomorrow and will report my progress here. It seems that i'm not so far from the result! Bruno Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi 568...@gmail.com ha scritto: Also, take a look at kamailio-advanced.cfg, there is

Re: [SR-Users] Help with sip balancer

First of all I'd suggest to use http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb guide in combination with http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html But, assuming your platform is behind NAT, you need: 1st. Use rtpengine instead of

[SR-Users] Help with sip balancer

Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +--- [asterisk1]

Re: [SR-Users] Help with sip balancer

Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already included. Also you can use LCR for routing calls to different providers, a simple guide can be found here http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ 2015-08-12 0:41 GMT+03:00 Alexandru Covalschi

Re: [SR-Users] Help with 407 Proxy Auth. Required

Specifically, after auth_check line add: xlog(The return code is $rc\n); Can add additional lines to view the values of other pseudovariables http://www.kamailio.org/wiki/cookbooks/4.0.x/pseudovariables On 3/9/2015 7:56 AM, Daniel-Constantin Mierla wrote: On 09/03/15 15:41, Agiftel wrote:

Re: [SR-Users] help with a new install

Hello, check the network traffic to see if the packages are sent to the sipcapture node. If kamailio is involved somehow, try to run it with debug=3 in kamailio.cfg Cheers, Daniel On 06/03/15 21:33, David Dunlap wrote: Hello, I am seeking help with a new test server. I have SIP messages

[SR-Users] Help with 407 Proxy Auth. Required

Hi all, i cannot understand where is the problem with this transaction: Kamailio ask for Proxy authorization and in the second INVITE credentials are present. Can you help me understand? Regards U 2015/03/09 15:03:42.191831 10.160.21.51:5060 - 10.160.20.18:5060 INVITE

Re: [SR-Users] Help with 407 Proxy Auth. Required

On 09 Mar 2015, at 15:07, Agiftel agif...@gmail.com wrote: Hi all, i cannot understand where is the problem with this transaction: Kamailio ask for Proxy authorization and in the second INVITE credentials are present. Can you help me understand? This is typical - it happens when the

Re: [SR-Users] Help with 407 Proxy Auth. Required

On 09/03/15 15:41, Agiftel wrote: Thanks Olle for reply but password is correct. Set the debug=3 in kamailio.cfg and check the log messages, you should get more hints for what part mismatches there. Also, after doing auth_check(), you can print the $rc to see the return code value. Cheers,

Re: [SR-Users] Help with 407 Proxy Auth. Required

Thanks Olle for reply but password is correct. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Help-with-407-Proxy-Auth-Required-tp136043p136045.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express

[SR-Users] help with a new install

Hello, I am seeking help with a new test server. I have SIP messages coming from a working FreeSwitch and getting posted into the sip_capture table. Homer web service is responding allowing me to manage log on accounts and other database settings however I do not see any packets in the web gui.

Re: [SR-Users] Help with dialog $dlg_var(cseq_diff)

Hello, $dlg_var(cseq_diff) is incremented after sending the invite out from failure route, being done when forwarding callback in dialog detects that the cseq value has to be incremented. I am going to test and see if there is an issue -- uac_auth() should set some internal flag to tell dialog

Re: [SR-Users] Help with dialog $dlg_var(cseq_diff)

Great! I will waiting for answer. If it needed I may make some tests. We building new system and want to use this technology insread of classic gateway. We will happy to cooperate with you for findinf issues and solve it as faster as we may. Thanks! 2014-11-03 20:03 GMT+04:00 Daniel-Constantin

Re: [SR-Users] Help with dialog $dlg_var(cseq_diff)

I just pushed a patch to master, can you try with it and if all is ok, then I will backport. Cheers, Daniel On 03/11/14 17:31, Yuriy Gorlichenko wrote: Great! I will waiting for answer. If it needed I may make some tests. We building new system and want to use this technology insread of

[SR-Users] Help with dialog $dlg_var(cseq_diff)

Hello. I need to increment CSeq value for INVITE with Auth params when use UAC_AUTH for outgoing calls to provider. Kamailio 4.2 may increment this using dialog module http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html Now I experements with this and var

Re: [SR-Users] Help to build a Kamailio SBC

Hello, the situation is: client - kamailio + rtproxy - asterisk - rtpproxy + kamailio - other client the idea is that rtpproxy has to proxy the whole rtp traffic between asterisk or other media proxy and the client. the strange problem is that my client which is receiving the call (csipsimple

[SR-Users] Help to build a Kamailio SBC

Hi to everyone, I’m trying to implement a SBC for my network based on Kamailio The base idea is putting Kamailio on a dual home machine (one public interface for clients and a private interface for media servers and database), using rtpproxy to handle the relaying from public to private

Re: [SR-Users] Help to build a Kamailio SBC

Hello, understanding the config file is going to take some time, so it is unlikely many will have the spare time for it. You have to provide more specific details, like the sip trace (ngrep on kamailio server sip ports) for the issue, presenting what are the parties involved in sending and

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

On 02/09/14 19:05, Alex Villací­s Lasso wrote: El 02/09/14 05:17, Daniel-Constantin Mierla escribió: If you get signling routed ok but no audio, then you have problems bridging rtp stream. Most probably you need to use rtpproxy (eventually with advertise address (there is a patch or use

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

If you get signling routed ok but no audio, then you have problems bridging rtp stream. Most probably you need to use rtpproxy (eventually with advertise address (there is a patch or use second parameter for rtpproxy_manage())) to bridge. I never used sip-natting in kernel, so I am not

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

El 02/09/14 05:17, Daniel-Constantin Mierla escribió: If you get signling routed ok but no audio, then you have problems bridging rtp stream. Most probably you need to use rtpproxy (eventually with advertise address (there is a patch or use second parameter for rtpproxy_manage())) to bridge.

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

On 29/08/14 23:58, Andres wrote: On 8/29/14, 1:42 PM, Alex Villací­s Lasso wrote: Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170: IP 198.58.101.75.5060 201.234.196.170.5060 INVITE

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

El 01/09/14 05:15, Daniel-Constantin Mierla escribió: On 29/08/14 23:58, Andres wrote: On 8/29/14, 1:42 PM, Alex Villací­s Lasso wrote: Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170:

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

El 01/09/14 10:50, Alex Villací­s Lasso escribió: El 01/09/14 05:15, Daniel-Constantin Mierla escribió: On 29/08/14 23:58, Andres wrote: On 8/29/14, 1:42 PM, Alex Villací­s Lasso wrote: Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here

Re: [SR-Users] Help analysing segmentation fault

Hi Both, In this case the system was using binaries from opensuse build service, so gcc I believe? Cheers, Charles On 28 Aug 2014 21:25, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 28/08/14 20:32, Jason Penton wrote: Hey Daniel, I am puzzled by how this could make any

[SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170: IP 198.58.101.75.5060 201.234.196.170.5060 INVITE sip:*43@201.234.196.170:5060 SIP/2.0 Via: SIP/2.0/UDP

Re: [SR-Users] Help analysing segmentation fault

Hey Daniel, I am puzzled by how this could make any difference? Could you explain? Is this dependent on the compiler used and whether or not void* arithmetic is allowed? Cheers Jason On Fri, Aug 22, 2014 at 1:17 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, can you try

Re: [SR-Users] Help analysing segmentation fault

Hello, On 28/08/14 20:32, Jason Penton wrote: Hey Daniel, I am puzzled by how this could make any difference? Could you explain? Is this dependent on the compiler used and whether or not void* arithmetic is allowed? void is incomplete type, of no defined data size, you cannot have: void

Re: [SR-Users] Help analysing segmentation fault

Yeah, would be cool to see what compiler Charles is using. Thanks Daniel On 28 Aug 2014 22:25, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 28/08/14 20:32, Jason Penton wrote: Hey Daniel, I am puzzled by how this could make any difference? Could you explain? Is this

Re: [SR-Users] Help analysing segmentation fault

Hi Daniel, The patch has tested OK so far. Regards, Charles On 22 August 2014 12:37, Charles Chance charles.cha...@sipcentric.com wrote: Thanks, Daniel. It can be hours, days or weeks between occurrences, but I will report back after a day or two initially then continue to monitor.

[SR-Users] Help analysing segmentation fault

Hi All, I wonder if some one could help me to diagnose a recurring issue? It happens at random times/intervals and under varying load. But always, just before the time of crash, I see the same critical error in log: CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in state

Re: [SR-Users] Help analysing segmentation fault

Hello, can you try this small patch? diff --git a/modules/pua_dialoginfo/pua_dialoginfo.c b/modules/pua_dialoginfo/pua_dialoginfo.c index 1e88a04..0f02b2b 100644 --- a/modules/pua_dialoginfo/pua_dialoginfo.c +++ b/modules/pua_dialoginfo/pua_dialoginfo.c @@ -347,7 +347,7 @@ struct str_list*

Re: [SR-Users] Help analysing segmentation fault

Thanks, Daniel. It can be hours, days or weeks between occurrences, but I will report back after a day or two initially then continue to monitor. Cheers, Charles On 22 Aug 2014 12:18, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, can you try this small patch? diff --git

[SR-Users] Help me to configure Kamailio on EC2, not able to connect through Jitsi

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=my-private-ip advertise=my-public-ip in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts

[SR-Users] Help me to configure kamailio on EC2

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=my-private-ip advertise=my-public-ip in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts

Re: [SR-Users] Help me to configure kamailio on EC2

Hi, Is your soft-phone packets reaching your kamailio server?, if so what happens to REGISTER packet?. On Tue, Jul 8, 2014 at 1:28 PM, Jayaraman, Kamalakannan kamalakannan.jayara...@pearson.com wrote: Hi, I had installed kamailio on my AWS EC2, by following the steps in

Re: [SR-Users] Help me to configure kamailio on EC2

You need to add user to Kamailio with username and password by using kamctl. Please try UDP on JITSI, if you still have problems please collect wireshark logs and send to us. Thanks, Veera On Tue, Jul 8, 2014 at 2:39 AM, Salman Zafar msalman...@gmail.com wrote: Hi, Is your soft-phone

Re: [SR-Users] Help with load balancing Kamalio based on DNS

Hello, what you can do is to route the messages to the other proxy if there is no contacts in the local location table, something like: if(!lookup(location) { if(src_ip==_THE_OTHER_KAMAILIO_IP_) { send_reply(404, Not found); exit; } $du =

[SR-Users] Help diagnosing non-response from incoming call

I have this setup for kamailio + asterisk, in which kamailio is supposed to listen on all ethernet interfaces on UDP port 5060, and will forward traffic from/to asterisk running on the same machine, and listening on localhost, udp port 5080. The scenario for the problematic call is somewhat like

[SR-Users] help with SIP Notify message, to make a endpoint (SPA504G and similar) to reboot.

Sorry for intruding, but this is puzzling me a lot. I followed the guide to make Kamailio work with asterisk and realtime, Kamailio version 4 and asterisk 11. (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ) up until now its working, and now want to get the phones

Re: [SR-Users] help with SIP Notify message, to make a endpoint (SPA504G and similar) to reboot.

On Thu, Mar 13, 2014 at 8:26 AM, Pedro Niño nino.pe...@gmail.com wrote: The other (ugly) option, is to remove the auth from the phone, for the Sip Provisioning, but that would leave and open door to a reboot attack without auth needed from any IP. And I dont like that option. This might not

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