Apparently the process of the installation did not setup even a default sip.conf file, and there were some other issues with the Kamailio.cfg file. Most of the issues were resolved by one of the other users in the group, and we were able to get the system running.
Thanks everyone for your assistance.
-----Original Message----- From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request@lists.sip-router.org Sent: Wednesday, December 29, 2010 11:49 AM To: sr-users@lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 75
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Today's Topics:
1. Re: Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database (Klaus Darilion) 2. Load Balancing (Tim King) 3. Re: Load Balancing (Alex Balashov) 4. fifo dp_translate (Borin) 5. Re: some issues with database initialization (berkeley/txt) (Noa Resare) 6. Re: some issues with database initialization (berkeley/txt) (Daniel-Constantin Mierla)
----------------------------------------------------------------------
Message: 1 Date: Wed, 29 Dec 2010 15:17:41 +0100 From: Klaus Darilion klaus.mailinglists@pernau.at Subject: Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database To: sr-users@lists.sip-router.org Message-ID: 4D1B4305.30504@pernau.at Content-Type: text/plain; charset=windows-1252; format=flowed
On 28.12.2010 22:07, Kurt Mullen wrote:
I am on my 20^th (I?m not kidding) attempt to successfully complete this tutorial.
I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & Asterisk on the same server as in the tutorial.
I have two SIP clients registered, but they are not able to call each
other.
No one answered my last two posts, so I hope someone can help this time.
If you can describe the problem (why can't they call each other) we can help you.
So, first you have to find out what is the problem. Use a packet sniffer (wireshark, ngrep) to capture the SIP traffic and analyze it (yes, if you want to use Kamailio you have to know at least the basics of SIP) for the problems.
regards klaus
------------------------------
Message: 2 Date: Wed, 29 Dec 2010 09:31:28 -0500 From: Tim King tim@compnetwork.net Subject: [SR-Users] Load Balancing To: sr-users@lists.sip-router.org Message-ID: AANLkTindx_r-RmTJSjTnpfdmLX8Nespzs_M9O0f6y385@mail.gmail.com Content-Type: text/plain; charset="iso-8859-1"
I have read many examples and it seems I have found several pieces of how to do this but no real example of how to achieve this.
I followed this setup guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
Performed this Kamailio install: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git
I have found the page for the dispatcher module here: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html
I also am curious if this dispatcher setup can be driven solely from the dispatcher mysql table or if the dispatcher.list file is required.
Can someone point me towards the proper documentation that truly show how to configure these features.
*SIP Routing Capabilities*
- - NAT traversal support for SIP and RTP traffic - load balancing with many distribution algorithms and failover support - flexible least cost routing - routing failover - replication for High Availability (HA)
* Load Based Load Balancing ** ------------------------- ** ** +----------+ ** | Kamailio | ** +----------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 1 | [Calls 1,3,7] ** | +--------------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 2 | [Calls 2,6] ** +--------------+ *
On 12/29/10 10:21 PM, Kurt Mullen wrote:
Apparently the process of the installation did not setup even a default sip.conf file, and there were some other issues with the Kamailio.cfg file.
The tutorial's focus is to show how to add kamailio to an existing Asterisk-based service, therefore servicing configuration of Asterisk was out of the scope.
Most of the issues were resolved by one of the other users in the group, and we were able to get the system running.
If you can point what is wrong with kamailio.cfg, I can get it fixed on the wiki page. The goal was to start from the default kamailio config file so everyone can do a diff and easily spot the changes -- in this way, people can update their configuration file by taking the additions and plugging to their server config.
Thanks, Daniel
Thanks everyone for your assistance.
-----Original Message----- From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request@lists.sip-router.org Sent: Wednesday, December 29, 2010 11:49 AM To: sr-users@lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 75
Send sr-users mailing list submissions to sr-users@lists.sip-router.org
To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request@lists.sip-router.org
You can reach the person managing the list at sr-users-owner@lists.sip-router.org
When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..."
Today's Topics:
1. Re: Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database (Klaus Darilion) 2. Load Balancing (Tim King) 3. Re: Load Balancing (Alex Balashov) 4. fifo dp_translate (Borin) 5. Re: some issues with database initialization (berkeley/txt) (Noa Resare) 6. Re: some issues with database initialization (berkeley/txt) (Daniel-Constantin Mierla)
Message: 1 Date: Wed, 29 Dec 2010 15:17:41 +0100 From: Klaus Darilionklaus.mailinglists@pernau.at Subject: Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database To: sr-users@lists.sip-router.org Message-ID:4D1B4305.30504@pernau.at Content-Type: text/plain; charset=windows-1252; format=flowed
On 28.12.2010 22:07, Kurt Mullen wrote:
I am on my 20^th (I?m not kidding) attempt to successfully complete this tutorial.
I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio& Asterisk on the same server as in the tutorial.
I have two SIP clients registered, but they are not able to call each
other.
No one answered my last two posts, so I hope someone can help this time.
If you can describe the problem (why can't they call each other) we can help you.
So, first you have to find out what is the problem. Use a packet sniffer (wireshark, ngrep) to capture the SIP traffic and analyze it (yes, if you want to use Kamailio you have to know at least the basics of SIP) for the problems.
regards klaus
Message: 2 Date: Wed, 29 Dec 2010 09:31:28 -0500 From: Tim Kingtim@compnetwork.net Subject: [SR-Users] Load Balancing To: sr-users@lists.sip-router.org Message-ID: AANLkTindx_r-RmTJSjTnpfdmLX8Nespzs_M9O0f6y385@mail.gmail.com Content-Type: text/plain; charset="iso-8859-1"
I have read many examples and it seems I have found several pieces of how to do this but no real example of how to achieve this.
I followed this setup guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
Performed this Kamailio install: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git
I have found the page for the dispatcher module here: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html
I also am curious if this dispatcher setup can be driven solely from the dispatcher mysql table or if the dispatcher.list file is required.
Can someone point me towards the proper documentation that truly show how to configure these features.
*SIP Routing Capabilities*
- - NAT traversal support for SIP and RTP traffic - load balancing with many distribution algorithms and failover support - flexible least cost routing - routing failover - replication for High Availability (HA)
- Load Based Load Balancing
** ------------------------- ** ** +----------+ ** | Kamailio | ** +----------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 1 | [Calls 1,3,7] ** | +--------------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 2 | [Calls 2,6] ** +--------------+
-
Daniel-Constantin Mierla -
Kurt Mullen