sr-users October 2009

sr-users@lists.kamailio.org

63 participants
105 discussions

[Kamailio-Users] How to take MAC address

by Chandrakant Solanki

Hi I want to know the MAC address of registered client on kamailio... suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and 192.168.1.4 from these 3 Client, i am registered successfully with 101, 102, 103(a)test.com But for additional authentication ... i also need MAC address for this Client. Is there any module, method, function or variable to take MAC address or is there any alternative for the above case. Waiting for favorable reply... -- Regards, Chandrakant Solanki

15 years

[Kamailio-Users] Pass calls from/to another realm to gateway

by Alexander

I've encountered one problem I can not solve :( The situation is following: we've got Kamailio working together with our own SIP platform. Our platform is about various call processing business logic and billing. All calls from SIP users should pass through our platform. For now we successfully can make and receive calls inside our domain - NAT is handled fine in most cases, instant messages are handled and so on. It looks like that: - Kamailio receives a call from caller - Call is redirected to our platform - Platform redirects call back to Kamailio and it looks for callee. I've tried to implement that logic. It seems to work in some cases, but I encounter one problem. Sometimes Kamailio can not forward a call to the platform via rewritehostport(). It tries to forward SIP request, but nothing happens - only retransmission handler is called. I mean, after call to t_relay(), I see this: Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... SIP requests seem to be correct, and configuration file works ok from time to time. The problem I've described happens when Kamailio receives a call from another domain (for example, from sipbroker). Inside our domain everything works fine and INVITE passes to our platform correctly. I've attached configuration file and debug output. Files were written for and by Opensips, but with Kamailio I encounter the same problem. Is this a problem with configuration file or something else?

15 years

[Kamailio-Users] roadmap to kamailio 3.0.0

by Daniel-Constantin Mierla

Hello, I am thinking of planning to release Kamailio 3.0.0 next Wednesday or day after (October 28 or 29). With some delays, testing is going fine in my side, still some bits to be done, here is what I collected: - taking in consideration $du updates in branch route - import fixes from sr_3.0 branch - complete core statistics and handling of drop reply I believe is feasible, in the worse case we can package it at respective date and allow a bit more time for testing, but we unfreeze code and can go for new development. I encourage people to start writing migration documents. Let's see other opinions as well. Cheers, Daniel

15 years

Re: [Kamailio-Users] How to take MAC address

by Zahid Mehmood

I know that the Polycom phones can be configured to add the MAC address in one of the headers (probably the user agent hdr). -- Zahid ----- Original Message ----- From: users-bounces(a)lists.kamailio.org <users-bounces(a)lists.kamailio.org> To: users(a)lists.kamailio.org <users(a)lists.kamailio.org> Sent: Wed Oct 28 02:58:05 2009 Subject: Re: [Kamailio-Users] How to take MAC address On Wednesday 28 October 2009 06:35:28 Chandrakant Solanki wrote: > Hi > > I want to know the MAC address of registered client on kamailio... > > suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and > 192.168.1.4 > > from these 3 Client, i am registered successfully with 101, 102, > 103(a)test.com > > But for additional authentication ... i also need MAC address for this > Client. > > Is there any module, method, function or variable to take MAC address or is > there any alternative for the above case. > > > Waiting for favorable reply... You will not have luck outsite "your proxy network" ... I mean ... that if your proxy is on 192.168.0.10/24 and your clients on any other network, you will always get's the same MAC, the mac of the 192.168.0.10/24 gateway. Your only hope is that SOME SIP devices add their MAC as part of some subscribe or register petition. I don't see any security increase on this, because MAC's could be easily spoofed. -- Raúl Alexis Betancor Santana Dimensión Virtual _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

15 years

Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error

by Daniel-Constantin Mierla

On 28.10.2009 11:23 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:50 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** > OUT route[8] > > what processing are you doing in kamailio for this case? Do > you do > same failure routing to another destination if B does not > answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > > > your config is big to troubleshoot it quick and easy. Definitely > you have call forwarding, what version of kamailio are you using? > the header of file says openser 1.2. > > I suggest you use ngrep/wireshark to watch the sip traffic and put > xlog() messages in the config to correlate and get hints about > what happesn. > > Cheers, > Daniel > > PS. Keep cc-ing to mailing list, there are people that can help > better or faster inc ase they use the sip wizard config. > > -- > Daniel-Constantin Mierla > * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin > * http://www.asipto.com/index.php/sip-router-masterclass/ > > > hi > > i m using kamailio-1.5.2-tls... > if you don't keep mailing list in the discussion thread, your messages will be ignored... Daniel

15 years

Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error

by Daniel-Constantin Mierla

On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] > > what processing are you doing in kamailio for this case? Do you do > same failure routing to another destination if B does not answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > your config is big to troubleshoot it quick and easy. Definitely you have call forwarding, what version of kamailio are you using? the header of file says openser 1.2. I suggest you use ngrep/wireshark to watch the sip traffic and put xlog() messages in the config to correlate and get hints about what happesn. Cheers, Daniel PS. Keep cc-ing to mailing list, there are people that can help better or faster inc ase they use the sip wizard config. -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

15 years

[Kamailio-Users] kamctl online

by Chandrakant Solanki

hello I would like to print user@domainname while i execute kamctl online command... Any Idea -- Regards, Chandrakant Solanki

15 years

[Kamailio-Users] Asterisk to Kamailio Call forwarding error

by Chandrakant Solanki

Hi Call Sequence is ... Client A/B register on Kamailio and both are registered successfully.. Firewall is stopped. Now... Client A call to Client B.. Client A => Kamailio => Asterisk => Kamailio => Client B Above is call sequence, when call goes from Asterisk to Kamailio ... then found below error.... ERROR:tm:t_forward_nonack: no branch for forwarding Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ERROR:tm:w_t_relay: t_forward_nonack failed Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] -- Regards, Chandrakant Solanki

15 years

Re: [Kamailio-Users] hung dialogs + no BYE

by Daniel-Constantin Mierla

On 27.10.2009 21:22 Uhr, Anders wrote: > That was exactly the problem Daniel - no BYE was ever sent from the > UA, so that's what we need to fix! > please keep cc-ing the mailing list. You can fix the problem by identifying the sip devices that do not send BYE. You are doing record routing, right? In 1.5, dialog can send BYE at call timeout -- does not help much you, but can close eventual open channels in gateways. The missing BYE happens when call is between two SIP phones? Or between sip phone and pstn gateway/media server? Cheers, Daniel > Thanks!! > > On Tue, Oct 27, 2009 at 4:16 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> On 26.10.2009 16:41 Uhr, Anders wrote: >> >>> Hi, >>> >>> I have two issues, and I think they are connected. The number of >>> Active Dialogs keeps growing - as if some of them are hung. Not all of >>> them, but some of them. At the same time, I have seen that from a >>> specific customer, there is no BYE in the 'acc' table in the >>> accounting. So, my conclusion - no BYE means it's not finished means >>> it's hung... - right? >>> >>> Any ideas where to look? >>> >>> >>> >> if you don't get a BYE in acc table then maybe was not sent and that keeps >> the dialog active. You can set a max time per call -- timeout -- for each >> dialog: >> >> http://kamailio.org/docs/modules/1.5.x/dialog.html >> >> However, is good to identify why BYE is not coming, maybe is a fraud attempt >> or a broken sip device. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

15 years

[Kamailio-Users] newbie question

by Luis Jimenez

Hello, list. Exactly what happens with a dialog (at accounting level) when i reload kamailio with dialogs in process? Thanks in advance. Luis Jimenez

15 years

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sr-users October 2009