November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_voicemail.c
by Archana Venugopan 12 Nov '12

12 Nov '12

Hi, I have got a doubt. In mod_voicemail.c script I see the below query "insert into voicemail_msgs(created_epoch, read_epoch, username, domain, uuid, cid_name, " "cid_number, in_folder, file_path, message_len, flags, read_flags, forwarded_by) " "values(%ld,0,'%q','%q','%q','%q','%q','%q','%q','%u','','%q','%q')", (long) switch_epoch_time_now(NULL), myid, domain_name, use_uuid, caller_id_name, caller_id_number, myfolder, file_path, message_len, read_flags, switch_str_nil(forwarded_by)); Want to know from where myid and domain_name are picked up. Initially I thought it was picking from voicemail_prefs but even if there are no entries for a caller in voicemail_prefs table it still insert the value. Please let me know on this. Mod_voicemail.c code is the same as present online. Regards, Archana

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(no subject)
by Jim Daley 12 Nov '12

12 Nov '12

I hope this is the right place to ask this question. I am running freeswitch using LUA. I'm doing outbound calling for a load tester. I have a number that when you call it you need to enter 2654# before the call get's answered. So I need a way of dialing DTMF digits before the other end answers. I tried to just sleep and send_dtmf after the session.ready but that doesn't work. Is there a way to do this. Sample code below...thanks if (session:ready()) then session:answer(); session:setVariable("CallCmp", "Script_dcetest1") session:setVariable("PromptFail", "Script_dcetest1") ; api = freeswitch.API(); session:sleep(1000); --status = session:getVariable("SilenceStatus"); session:execute("wait_for_silence", "200 30 20 20000"); -- -- Continue in english, press 2 status = session:getVariable("SilenceStatus"); -- This will get past langauge and main menu prompts if((status == "unset")and(session:ready())) then session:execute("send_dtmf", "2@100"); --@ determines dtmf length, enter NPI

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Multiple profiles on a single domain
by Emrah 12 Nov '12

12 Nov '12

Hi Freeswitchers, I had to start an additional profile on port 5070 because my Verizon MiFi blocks 5060 completely. For now, 5070 allows me to have a call for 120 seconds… Than the RTP session drops. Here is how I started my new profile. Sounds too simple to be true! cp internal.xml internal-5070.xml Edit internal-5070.xml to change the port number Run reloadxml followed by sofia profile internal-5070 start… I can register and make short calls, but I do not see my registration when I issue a sofia_contact username@domain and incoming calls do not work. I did not edit anything else in internal-5070.xml and do not force a registration domain either. Sofia status does not show me an additional domain name being created. The documentation is a bit unclear to me on how FS finds it's domains… I am sure it is something with the automatic parsing parameter at the top of my sofia profiles, but it still doesn't click… any hint will be enormously appreciated. Thanks! E

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general question about phone provisioning
by Abaci 12 Nov '12

12 Nov '12

This question is not specific to FreeSWITCH, just a general question that I would like to get feedback from other FreeSWITCH users. I'm thinking of setting up phone provisioning via http, my question is how to make this setup secure. say my provisioning server will listen on https://myserver.com and a phone with the mac address 00-15-65-22-F4-23 will try to pull the config as https://myserver.com /00156518425Dhow do I prevent hackers from trying to get config files using a brute force attack. is there any standard way of securing against these types of attacks?

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voicebox email notifications on Debian with exim4
by Daniel-Constantin Mierla 12 Nov '12

12 Nov '12

Hello, I got into the crashing issue for freeswitch voicemail email notifications on Debian with exim4, reported at: * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings My solution was to use msmtp alongside exim4 -- msmtp is used to relay the emails to the local exim4 instance. All seems to work ok, I wonder if anyone else is using same approach and got any issues or there are other solutions found meanwhile for debian+exim4 combination. I plan to put more details there, once the registration process for my account on the wiki is completed. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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Nested contexts emulation
by Eugene Prokopiev 11 Nov '12

11 Nov '12

Hi, I can include one context to another. It is very convenient to avoid duplication of the configuration code. With FreeSWITCH it is also quite in demand feature for the same extensions in different contexts for different profiles. But the only way to avoid duplication is to have one big context and block certain extensions by conditions for different profile. Can I do this by another way? -- Regards, Eugene Prokopiev

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Re: [Freeswitch-users] Error 606 (USER_NOT_REGISTERED) but in fact he is...
by Mi Ke 11 Nov '12

11 Nov '12

Anton, thanks for your help - the problem was pre-process directive in config pointing to internal profile by default while I was using external. Everything works now. ----- Original Message ----- From: Anton Kvashenkin Sent: 11/09/12 06:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error 606 (USER_NOT_REGISTERED) but in fact he is... Which soifa profile do you use for registrering this user. Can you post sofia_contact */thatuser@domain. Also, do not forget abount dial_string. 2012/11/9 Mi Ke < mi.ke(a)null.net > Hi, I have an IP phone behind NAT registered to also NATed FreeSWITCH (EC2): 10.10.10.107 (phone) -> 111.111.111.111 (NAT) -> internet -> 222.222.222.222 (NAT) -> 10.210.250.135 (FS) Registration part of sip trace looks OK (?): tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fdcf8004430): events IN tport.c:2845 tport_recv_event() tport_recv_event(0x7fdcf8004430) tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x7fdcf8004430) msg 0x7fdcf8010980 from (udp/http://10.210.250.135:5060 ) has 702 bytes, veclen = 1 tport.c:3004 tport_deliver() tport_deliver(0x7fdcf8004430): msg 0x7fdcf8010980 (702 bytes) from udp/http://111.111.111.111:5060/sip next=(nil) nta.c:2803 agent_recv_request() nta: received REGISTER sip:222.222.222.222 SIP/2.0 (CSeq 49512) nta.c:3090 agent_check_request_via() nta: Via check: received=111.111.111.111 nta.c:3161 agent_aliases() nta: canonizing sip:222.222.222.222 with contact nta.c:3002 agent_recv_request() nta: REGISTER (49512) going to a default leg 2012-11-09 01:14:34.833161 [ERR] sofia_reg.c:1613 DELETE PRESENCE SQL: delete from sip_presence where sip_user='1000' and sip_host='222.222.222.222' and profile_name='external' and open_closed='closed' nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_respond nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_destroy tport.c:3238 tport_tsend() tport_tsend(0x7fdcf8004430) tpn = UDP/http://111.111.111.111:5060 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x7fdcf8004430): not found by name UDP/http://111.111.111.111:5060 nta.c:6678 incoming_reply() nta: sent 200 OK for REGISTER (49512) Here registration LED on the phone goes green, phone gives me a dialing tone i.e. behaves like registered. show registrations looks OK: reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 1000,222.222.222.222, 6a7735fa-6f81e64c(a)10.10.10.107 ,sofia/external/sip:1000@10.10.10.107:5060;fs_nat=yes;fs_path=sip%3A1000%40111.111.111.111%3A5060,1352420589,111.111.111.111,5060,udp,domU-12-31-39-09-F5-79, sofia status profile external ... REGISTRATIONS 1 Meanwhile when incoming call hits my dialplan (simple bridge to user/1000@domain) I'm getting ERR 606 (User not registered) and here is what list_users gives to me: userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number 1000|default|10.210.250.135|default|error/user_not_registered||Mike|1000 The question is - what am I doing wrong? Thanks in advance for all yours hints Mike _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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are there any high performance rtmp dispatcher solution, comparing to the opensips/Kamailo (for SIP)?
by zhi sun 10 Nov '12

10 Nov '12

hi guys, i have an opensips as SIP dispatcher and loadbalance of several freeswitch (behind). For now, if the end-users use flash rtmp web phone to make a call, the opensip cannot handle the rtmp request, although the freeswitch has mod_rtmp. in this situation, could i use a freeswitch at the front of opensip, and woks only as rtmp proxy as below? FS (rtmp dispatcher) --> opensips --> FSs if not, should i use the 3rd party rtmp gateway/dispatcher to replace the opensips? are there any high performance rtmp dispatcher solution, comparing to the opensips/Kamailo (for SIP)? i am new to this area, thanks for any suggestions. -iamsyt

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mod_voicemail error
by Michael Cole 10 Nov '12

10 Nov '12

Nov 10 10:00:46 37 freeswitch[1857]: 2012-11-10 10:00:46.019811 [ERR] mod_voicemail.c:3311 Error creating /usr/storage/voicemail/default/137.220.92.81/1002 The user freeswitch has access to the directory read and write. I even changed the ACL on the directory to be RWX for all users but still getting the error. If I run the entire freeswitch as root then no problems but that is a complete no no .. Does anyone have an idea where i'm going wrong? Regards Michael

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Voicemail to email with exim
by Prasd D 10 Nov '12

10 Nov '12

I am using using exim with eximcompat.sh way (according to) http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings but i am not getting email. I don't see anything referring to email in the debug log also. Can anyone tell how to get this working ? Is there a way to make it work such that there's a reminder sent to the recipient with no attachment but with a link to the voicemail file ? -- Thanks, Prasd

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