[SR-Users] Kamailio with Asterisk
Hi all, I'm wondering if someone can help me get a little unstuck. I followed the guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And I now have kamailio running, but I have two problems. Number 1 When I do kamctl add user password and make two users, they register fine. But any attempt to call between the users times out. Number 2 Anytime I have my kamailo.cfg file begin as #!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_ASTERISK I am unable to start kamctl. I get the following error. ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed If I run kamailio I get ERROR: bad config file (3 errors) The moment I remove the define lines at the start, kamctl starts fine, but I can't authenticate with asterisk. Any ideas? Thanks Chirag ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with Asterisk
Hello Chirag, Ad 1. When making a test call do an ngrep on the kamailio: ngrep -d any -qt -W byline port 5060 That information can help you to do more troubleshooting. Ad 2. Enable the debug in the kamailion config : #!define WITH_DEBUG Then start kamailio again, you will see a lot of information that will show why kamailio wont start. You could also show the config as you have it now, it's hard to provide any guidance with just the info you are providing... Yes, I know, there are kamailio guru's that have enough on just two words, but there are only a few guru's. Good luck. Gertjan - Oorspronkelijk bericht - Van: "Chirag Desai" Aan: sr-users@lists.sip-router.org Verzonden: Maandag 27 januari 2014 20:33:42 Onderwerp: [SR-Users] Kamailio with Asterisk Hi all, I'm wondering if someone can help me get a little unstuck. I followed the guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And I now have kamailio running, but I have two problems. Number 1 When I do kamctl add user password and make two users, they register fine. But any attempt to call between the users times out. Number 2 Anytime I have my kamailo.cfg file begin as #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK I am unable to start kamctl. I get the following error. ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed If I run kamailio I get ERROR: bad config file (3 errors) The moment I remove the define lines at the start, kamctl starts fine, but I can't authenticate with asterisk. Any ideas? Thanks Chirag ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with Asterisk
If you have your kamailio.cfg in /etc/kamailio/ you could get a closer reason why kamailio isn’t starting by: kamailio -cf /etc/kamailio/kamailio.cfg and do a tail -n100 /var/log/syslog Op 28-jan.-2014, om 14:49 heeft Gertjan Wolzak het volgende geschreven: > > Hello Chirag, > > Ad 1.When making a test call do an ngrep on the kamailio: ngrep -d any > -qt -W byline port 5060 > That information can help you to do more troubleshooting. > > Ad 2.Enable the debug in the kamailion config : #!define WITH_DEBUG > > Then start kamailio again, you will see a lot of information that > will show why kamailio wont start. > > You could also show the config as you have it now, it's hard to provide any > guidance with just the info you are providing... > Yes, I know, there are kamailio guru's that have enough on just two words, > but there are only a few guru's. > > Good luck. > > Gertjan > > Van: "Chirag Desai" > Aan: sr-users@lists.sip-router.org > Verzonden: Maandag 27 januari 2014 20:33:42 > Onderwerp: [SR-Users] Kamailio with Asterisk > > Hi all, > > I'm wondering if someone can help me get a little unstuck. > > I followed the guide here: > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb > > And I now have kamailio running, but I have two problems. > > Number 1 > When I do kamctl add user password and make two users, they register fine. > But any attempt to call between the users times out. > > Number 2 > Anytime I have my kamailo.cfg file begin as > #!KAMAILIO > > #!define WITH_MYSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > #!define WITH_ASTERISK > > I am unable to start kamctl. I get the following error. > ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed > > If I run kamailio I get > ERROR: bad config file (3 errors) > > The moment I remove the define lines at the start, kamctl starts fine, but I > can't authenticate with asterisk. > > Any ideas? > > Thanks > > Chirag > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio with asterisk for outbound calls
Hi All Kamailio Experts, I have configured Kamailio (kamailio 3.1.5) as media server. All things are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound Calls. For this purpose i have followed the web page : http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. In this page, some points are not clear for me , as given below: (1) In case you use *sipregs* you have to create a record for each extension where to set the 'name' to value of 'name' from *sipusers*. The rest is populated by Asterisk from registrations. (2) Be sure you configure Asterisk *to not authenticate* SIP requests coming from Kamailio. I am not sure that my local users chat is working through kamailio or asterisk, who is used for authorization. Any specific Web page to correct the issue will highly appreciated according to my scenario. Kindly guide me. Thanks in advance. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.tha...@loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer - This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio with Asterisk 13 PjSip Realtime
Hi, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripling out authentication and registration from asterisk and solely letting Kamailio handle it. In order to do this would I be correct in assuming I would have to use the asterisk database rather than the Kamailio database? I've compared the two and the table structures are very different. If I use the asterisk database I guess asterisk still needs to be responsible for handling authentication, registration and writing the contacts to the database. If I use the Kamailio database how would I dial an extension from asterisk, because as far as I can fell asterisk will have no idea who is registered or how to find them (contact details). Maybe I'm over thinking something simple, or maybe I'm not. Either way I would love your thoughts on how this could be done. Kind regards, C ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with asterisk for outbound calls
On 22.08.2012 14:26, Vijay Thakur wrote: Hi All Kamailio Experts, I have configured Kamailio (kamailio 3.1.5) as media server. Kamailio is a SIP proxy, not a media server. Maybe you mean that you are using Kamailio with rtpproxy as media relay. All things are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound Calls. For this purpose i have followed the web page : If you wan to you Asterisk as PSTN gateway only, then there is no need to follow this tutorial. This tutorial makes strong integration of Kamailio and Asterisk. For PSTN gateway functionality there is no need to integrate Kamailio and Asterisk - just configure Asterisk as gateway and forwards PSTN calls from Kamailio to Asterisk (and vice versa) http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. In this page, some points are not clear for me , as given below: (1) In case you use *sipregs* you have to create a record for each extension where to set the 'name' to value of 'name' from *sipusers*. The rest is populated by Asterisk from registrations. > (2) Be sure you configure Asterisk *to not authenticate* SIP requests coming from Kamailio. I am not sure that my local users chat is working through kamailio or asterisk, who is used for authorization. What do you mean with "not sure"? For instant messaging between users there is no need to use Asterisk. In above setup the authentication is done by Kamailio only. regards Klaus Any specific Web page to correct the issue will highly appreciated according to my scenario. Kindly guide me. Thanks in advance. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail :vijay.tha...@loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web:www.loopmethods.com LOOP Disclaimer - This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with asterisk for outbound calls
Thanks for clearing the doubts. You are very right, i am using kamailio as Media Relay. Can you send me some specific document URL, from where i can configure Asterisk as PSTN Gateway. Can we set Kamailio and Asterisk in one server. Thanks in advance. Vijay Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote: On 22.08.2012 14:26, Vijay Thakur wrote: Hi All Kamailio Experts, I have configured Kamailio (kamailio 3.1.5) as media server. Kamailio is a SIP proxy, not a media server. Maybe you mean that you are using Kamailio with rtpproxy as media relay. All things are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound Calls. For this purpose i have followed the web page : If you wan to you Asterisk as PSTN gateway only, then there is no need to follow this tutorial. This tutorial makes strong integration of Kamailio and Asterisk. For PSTN gateway functionality there is no need to integrate Kamailio and Asterisk - just configure Asterisk as gateway and forwards PSTN calls from Kamailio to Asterisk (and vice versa) http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. In this page, some points are not clear for me , as given below: (1) In case you use *sipregs* you have to create a record for each extension where to set the 'name' to value of 'name' from *sipusers*. The rest is populated by Asterisk from registrations. > (2) Be sure you configure Asterisk *to not authenticate* SIP requests coming from Kamailio. I am not sure that my local users chat is working through kamailio or asterisk, who is used for authorization. What do you mean with "not sure"? For instant messaging between users there is no need to use Asterisk. In above setup the authentication is done by Kamailio only. regards Klaus Any specific Web page to correct the issue will highly appreciated according to my scenario. Kindly guide me. Thanks in advance. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail :vijay.tha...@loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web:www.loopmethods.com LOOP Disclaimer - This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with asterisk for outbound calls
On 23.08.2012 08:31, Vijay Thakur wrote: Thanks for clearing the doubts. You are very right, i am using kamailio as Media Relay. Can you send me some specific document URL, from where i can configure Asterisk as PSTN Gateway. There is no such document. But configuring a PSTN gateway is already in the default configuration file. Just search in the deafult configuration file for "WITH_PSTN". Can we set Kamailio and Asterisk in one server. Yes, thats no problem. Either use 2 IP addresses on the same server, one for Kamailio and one for Asterisk, or use the same IP address and different ports. regards Klaus PS: If you are building a public SIP service it is a good idea to not use the default port 5060 to get rid of SIP port scanners. Thanks in advance. Vijay Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote: On 22.08.2012 14:26, Vijay Thakur wrote: Hi All Kamailio Experts, I have configured Kamailio (kamailio 3.1.5) as media server. Kamailio is a SIP proxy, not a media server. Maybe you mean that you are using Kamailio with rtpproxy as media relay. All things are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound Calls. For this purpose i have followed the web page : If you wan to you Asterisk as PSTN gateway only, then there is no need to follow this tutorial. This tutorial makes strong integration of Kamailio and Asterisk. For PSTN gateway functionality there is no need to integrate Kamailio and Asterisk - just configure Asterisk as gateway and forwards PSTN calls from Kamailio to Asterisk (and vice versa) http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. In this page, some points are not clear for me , as given below: (1) In case you use *sipregs* you have to create a record for each extension where to set the 'name' to value of 'name' from *sipusers*. The rest is populated by Asterisk from registrations. > (2) Be sure you configure Asterisk *to not authenticate* SIP requests coming from Kamailio. I am not sure that my local users chat is working through kamailio or asterisk, who is used for authorization. What do you mean with "not sure"? For instant messaging between users there is no need to use Asterisk. In above setup the authentication is done by Kamailio only. regards Klaus Any specific Web page to correct the issue will highly appreciated according to my scenario. Kindly guide me. Thanks in advance. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail :vijay.tha...@loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web:www.loopmethods.com LOOP Disclaimer - This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
