March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

suggestions to application
by roberto 11 Mar '09

11 Mar '09

Hello, I have an app write in php running with asterisk for calling cards, i would like to migrate everything to freeswitch, but i think that fs has almost everything that a calling card need read to use.. I´m looking for suggestion on some features like: Accounting Rate Billing Route Please someone could give me any suggestion on wich program that i should use. Thanks, -- "Without love, we are birds with broken wings." Morrie

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Waiting for playing a file as early media
by Helmut Kuper 11 Mar '09

11 Mar '09

Hello, I'm looking for a way to play a file exactly once in early media state of a call and then do e.g. a 302 reply. When I try it via ring_ready and ringback The playback immediately stops, when I send a respond. regards Helmut

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Where does the FS keep the call-id list?
by shawn lin 11 Mar '09

11 Mar '09

Hi all, I am new to FreeSwitch, I have a question when using SIPp <-> FreeSwitch <-> SIPp. The SIPp UAC create many calls and the SIPp UAS does reveive them. I wonder if anyone can tell me, where does the FreeSwitch keep the call-ids? Is there a list to contain all the call-id? If there is a list, how can I found it? Best Regards! Shawn Lin

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IRC pre release party
by Anthony Minessale 10 Mar '09

10 Mar '09

If you have nothing better to do drop by IRC We are up to 193 users and about to cross 200 for the first time. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Missing Diversion header in INVITE after 302 reply
by Helmut Kuper 10 Mar '09

10 Mar '09

Hello, following scenario: -Phone A is redirected unconditionally to phone C -Phone B calls A -Phone A replys with 302 and Dieversion header -FS detects the 302 and sends out a new INVITE to C I found that FS doesnt' include the received diversion sip header into the new INVITE sent to phone C. Is there a way to configure FS so that diversion header are included? Additionally: Is there a way to inform phone A about the diversion header, so that phone A get display a hint to user? regards Helmut

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FS stopped working when NIC connection bounced.
by Richard Lamkin 10 Mar '09

10 Mar '09

Hi All, FS stopped working when NIC connection bounced. Q1- Has anyone an explanation of what happened ? Q2 - Is there a way to configure FS not to flip to another IP on a lost network connection. Q3 - Should an IP changed event be logged at a higher level e.g. [CRITICAL] ? NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped back to NIC2. When it flipped back no SIP connections were restored. All PBX phones lost registration and my gateways to remote switches did not recover. FS is running but not responding to CLI or SIP. There is nothing logged in the Windows event viewer at the time of the incident. Sorry but I was not running Wireshark when this happened. ========== Log extract ======== 2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP change detected [172.22.240.156]->[192.168.1.1] []->[] 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write lock external 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting for worker thread 2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write lock internal 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting for worker thread 2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP change detected [192.168.1.1]->[172.22.240.156] []->[] 2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions There were no more logged events after 2009-03-09 21:11:21. The log file was checked at 2009-03-10 08:50 . All phones had lost registration permanently I tried the CLI with "sofia status" this failed to return anything and the CLI was no longer responsive. I did a CTR-C just to see what was alive [ VS2008 trapped the break and I selected continue] 2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 4 seconds 1 heartbeat (core) Still no CLI. I the killed the FS, restarted and it worked as normal ==== Background; I'm testing Freeswitch on Windows XP with Release 1.03 from the source release tar ball, running in VS2008 debug mode. The PC has two NIC's 1 - 192.168.1.1 static 2 - 172.22.240.156 DHCP I'm using Nortel switches CS1K and CS2K for my upstream gateways and Linksys SPA942 for PBX clients. I'm only using SIP <-> SIP and all sip connections are via 172.22.240.156, There are no SIP devices on 192.168.1.1. It is unlikely the event was caused by a DHCP lease renew. DHCP lease details for 172.22.240.156. Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28 Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28 === Regards Richard Lamkin PS: I am a new to FS and I am very enthusiastic about it potential. ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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Re: [Freeswitch-users] OpenZap and Sangoma A500 BRI card
by Sergey Kirillov 10 Mar '09

10 Mar '09

> > I'm no BRI expert but it looks to me like your wanpipe is set up for > E1/EuroISDN. Where did you get this setup information? > -MC > It is autoconfigured by wancfg

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Re: [Freeswitch-users] OpenZap and Sangoma A500 BRI card
by Sergey Kirillov 10 Mar '09

10 Mar '09

Is there a development version that I can check now? On Mon, Mar 9, 2009, Anthony Minessale<Anthony Minessale> wrote: > it's not released yet, > > please wait for the announcement that it has been completed sometime in the > next week or 2.

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FS and Billing, AGI emulator
by Diego Viola 10 Mar '09

10 Mar '09

Hi, I wanted to use A2Billing on FS, but I noticed it uses some AGI stuff for dialling and to check how much credit the user has, etc. I heard you could use A2B by just importing the FS CDR data into it, but that wont work, so I come to the conclusion that I have no way of doing billing on FS yet. There is ASTPP but that's not complete yet, and I heard vague comments of doing an AGI emulator on top of the event socket on FS, how hard that would be? Is there a possibility to do that? How much would it cost? Thanks, Diego

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Script parsing a TPORT_DUMP sip log file to Mysql
by seven 10 Mar '09

10 Mar '09

Hi all, I wrote a ruby script. it works for me. The script is in <FS>/scripts/ contrib/seven/sip/. All ideas and suggestions are welcome! Comment in script: Now and then we need to look at sip traces to see want happened on a failed call. There are lots of ways to monitor sip messages. However, not all of them are convinient as we want. Let's say a simple example: FreeSWITCH :> originate sofia/gateways/carrier1/5550000|sofia/gateways/ carrier2/5550000|sofia/carrier3... It's hard to tell what happend if the call fails. Because it's different sip sessions. The idea is to group them in one super session and see what happend. I do this by adding an arbitary sip header to do cross reference. And by parse the sip messages to a DB we can easily show it as html. I even can build a simple graph based on the DB data: http://skitch.com/seven1240/b8xj2/voip-master-idapted You can easily add a sip header to INVITE by(I use x_interaction): FreeSWITCH :> originate {sip_h_x_interaction=TEST0001}sofia/ gateways/..... And I can get all the messages from DB: SELECT * FROM `sip_messages` WHERE (call_id IN (SELECT distinct call_id FROM sip_messages WHERE x_interaction = 'TEST0001')) ORDER BY created_at; There are two aproches to get sip packets: 1) tcpdump/tshark 2) FreeSWITCH I use the second. Note, there is no way to actually get sip messages from FS currently, but sofia-sip has the ability to log all sip messages to a disk file by using TPORT_DUMP And you can use this script to parse them to a DB. I know it hurt performance, but we don't have tons of traffic and you know there are only 5-10 messages for each sip call. While we get about 1G bytes each day in the sip log, most of them are OPTIONS/NOTIFY etc. I filtered them before inserting to DB, but it will be better if sofia- sip can filter that :) The script will monitor the log file and parse and insert to DB in real time. It's written in the Ruby on Rails framework, however, I think it can run out of Rails with or without modification. But you still need ruby and rubygems if you want to use it. on Ubuntu/Debian # apt-get install ruby rubygems # gem install mysql file-tail yaml It's just an idea, you may like to write your own tools to parse the sip log file. Also the log file need to be rotated regularly. And I think it maybe possible to store the log file on a memory disk, whatever... :) Best -Seven.

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