- FreeSWITCH-users - FreeSWITCH Mailing Lists

Issue in fifo while running it using outbound socket with async
by Abubacker siddiq 17 Jun '10

17 Jun '10

Dear all , -- BEST REGARDS N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23

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what's the cause of ozmod_isdn.c:1839 D-Chan Read Error!
by Tony Tin 17 Jun '10

17 Jun '10

Hi, I've Digium TE220p and 4ess ISDN with openzap native pri stack. I found there are a lot of errors as below in the freeswitch.log. It seems that it doesn't affect the operation of system at the beginning, but finally freeswitch stop answering call. Could anyone help me. Thanks. 1. What's the cause of this error message. 2. How to avoid. 2010-06-16 10:07:04.945026 [ERR] ozmod_isdn.c:1839 D-Chan Read Error! Regards, Tony

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Call disconnect response from FreeSWITCH
by arnuld uttre 16 Jun '10

16 Jun '10

I just started using FS (FreeSWITCH). I am connecting to it through telnet as client and giving it originate command to make calls to some SIP phone. Its placing calls without any problems but FS does not send any info when a call completes so I am little confused on how to know that a call is completed. A typical session is shown down here. FS tells immediately that the call is picked up by the person on other side but no information on when a call is completed. Any idea on how to know that a call is completed so that I can free up my malloc()ed memory. [arnuld@dune ~]$ telnet 192.168.0.222 8021 Trying 192.168.0.222... Connected to cobra (192.168.0.222). Escape character is '^]'. Content-Type: auth/request auth **** Content-Type: command/reply Reply-Text: +OK accepted api originate sofia/internal/105@192.168.0.228:5060 1111 Content-Type: api/response Content-Length: 41 +OK 19d962e3-9fa1-4f26-ba8c-0b9ac6afadc1 exit Content-Type: command/reply Reply-Text: +OK bye Content-Type: text/disconnect-notice Content-Length: 67 Disconnected, goodbye. See you at ClueCon! http://www.cluecon.com/ Connection closed by foreign host. [arnuld@dune ~]$ -- http://uttre.wordpress.com/2008/05/14/the-lost-love-of-mine/

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vars.xml for Spain
by luixsansan＠hotmail.com 16 Jun '10

16 Jun '10

Hello Reading the vars.xml file I found the following defaults  <X-PRE-PROCESS cmd="set" data="call_debug=false"/> <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/> <X-PRE-PROCESS cmd="set" data="default_areacode=918"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/> <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440.0,480.0)"/> <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440.0,0.0)"/> <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425.0,0.0)"/> <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425,0)"/> <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> I will assume default_areacode=91 for Madrid although the area code is added to all numbers in Spain and I wonder if it is better to leave this field blank, default_country will be ES but which is the es-ring, bong-ring and sit for Spain? Thank you very much for your help. Luis.

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Javascript Database Timestamp and Date()
by David Swardstrom 16 Jun '10

16 Jun '10

I am working with Javascript and Freeswitch to emulate some Conferencing capability that the company I am working for has in production using other products. One thing that I need to do is to access a timestamp from a database in JS. In pgAdmin for Postgres this is a "timestamp without time zone". In JS there is a data type called Date. Question: If I read a row of the database which includes this timestamp field, what should I put it into? * Can I put it directly into a Date data type? * Does it have to come out as a string or a number? * If it is a number can I then put it into a Date? * If it is a string, what is the format: Is it: "YYYY-MM-DD HH:MM:SS.ffffff"? I looked in mod_spidermonkey_odbc.c pdbc_next_row() and it appears to call SQLFetch() and return the result. But information on a row contains both a value but it's type, so what will the interpreter do? If anyone knows, let me know, otherwise I will incorporate a way to test this into my code and let everyone know (may also add this to some Wiki). Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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Your Security Best Practices
by Code Ghar 15 Jun '10

15 Jun '10

Recently I have seen many attempts to break in to VoIP switches I work with. So I thought this might be a good time to discuss everyone's security best practices. If you could relate them to how you implemented them in FreeSWITCH, it would give us beginners good ideas on how to follow the principle of "security first". Let me start off the discussion with some practices we use and have seen in carriers we deal with. Since we only do carrier-to-carrier stuff, it is easy to restrict the IPs where we accept traffic from. This is done using either a dedicated firewall (we use Cisco and pfSense) or host-based (since we use Linux, iptables it is). We restrict ingress traffic on port 5060 (SIP) to trusted sources only. Since a lot of carriers are starting to use re-invites and most do not provide ranges of media IPs they will use, we leave open all UDP ports used by our servers for RTP, usually between 10,000 and 50,000. In this situation, how well does FreeSWITCH cope with attacks? For example, if RTP ports are 10,000 to 20,000, does FS recognize invalid RTP packets and knows how to deal with them? Another scenario is user registration of softphones or ATAs, etc. A practice we use in our lab environment (we will soon be putting servers in production) is to use lengthy passwords, at least 15 characters, mixing alpha-numeric characters. We also do not use 4-digit extensions (default in FS) but use 6-digit. It wasn't started as a security measure but turned out that many attackers we saw usually try 4-digit extensions. So we are safer today (maybe not so in the future when attackers wise up). How do you manage your users, extensions, registrations, etc., from a security aspect? In the same context, we recently saw a flood of invalid user registration requests, at close to 60-80 requests per second, for a sustained period of less than a minute. One proprietary solution we use was unable to handle this flood of requests and dropped 70% of its call before the attacks stopped and it was able to recover. Let's say we use top-of-the-line servers to run FS, maybe two quad-cores with 16GB memory, then should it increase the ability of FS to deal with such unwanted floods? I assume FS is configured with default settings. The real question is: does throwing hardware at the problem solve the issue for FS? And even if it does, is there anything we can do at the FS level to allow it to handle even more flooded traffic on its own, without relying on superior hardware? Then come Man in the Middle attacks. So far I haven't seen a real push in the industry to mitigate these. One common practice is to use prefixes to identify traffic from a source. For example, we can use 999912125550000 where 9999 identifies us to the egress gateway and the rest is the number to call. Of course, since signaling is in plain text someone can analyze this data and then start using the same prefix to send traffic to our egress gateway/carrier. They believe the traffic is from us and charge us for the calls. A second option is to use SIPS but I haven't seen it being used at all. Of course, now we are talking about using TCP instead of UDP, bringing its own advantages and disadvantages. A third option I am starting to see if the use of IPSec tunnels for signaling and RTP is exchanged outside of the tunnel. This works well for carrier-to-carrier traffic but cannot be deployed widely cost-effectively if you do home user to service provider traffic. If you have dealt with any one or all of these scenarios, or have come across others, please share your experiences and solutions with all of us. Of course, if you can also advise how to implement your security best practices in FS it would be just wonderful.

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New provider based on asterisk instructions (xnet/worldxchange in New Zealand)
by Fraser Redmond 15 Jun '10

15 Jun '10

Hi, I'm trying to set up a new provider, based on the instructions they've provided for Asterisk. I'm primarily interested in inbound calls for now. I've set up new providers a couple of times before, but my guesswork isn't working out this time. I'm getting a 404 error. The instructions say: Edit your Sip_additional.conf file and add the following: [VFX] type=peer fromuser=DIDnumber host=pan.wxnz.net insecure=invite,port canreinvite=no nat=yes secret=PASSWORD username=USERID register=DIDnumber:PASSWORD:USERID@pan.wxnz.net/DIDnumber I suspect the problem is to do with the register line, any ideas on how to set that up in FreeSwitch? Or any other suggestions of settings I ought to try changing? Cheers, Fraser

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Re: [Freeswitch-users] Question about execute_on_answer
by Davide Colombo 15 Jun '10

15 Jun '10

Hi, using group_confirm_file and group_confirm_key i have only one choice, connect called party with calling party (1 dtmf maps with group_confirm_key). What i'd like to do when called party answers, it's: press 1 to connect with calling party, press 2 to hangup calling party, press 3 move calling party to voicemail. Additionally, making parellel dial in bridge, called phones stop to ring only when one of them digits the group_confirm_key and not (my purpose) when one of them answers. These are the reasons to transfer called channel to an ivr context using "execute_on_answer" variable. Best Regards ----- Messaggio originale ----- Da: "Michael Collins" <msc(a)freeswitch.org> A: freeswitch-users(a)lists.freeswitch.org Inviato: Martedì, 15 giugno 2010 0:37:18 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Aha! You are trying to do some sort of answer verification, correct? You do know that FS already has this feature, correct? Check it out: http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_varia… Make sure that you aren't duplicating functionality before continuing. -MC On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" < msc(a)freeswitch.org > A: freeswitch-users(a)lists.freeswitch.org Inviato: Lunedì, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: <action application="export" data="nolocal:execute_on_answer=transfer myivrpresentation XML default"/> <action application="bridge" data="user/1019@${domain_name}"/> When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000(a)fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ 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 _______________________________________________ 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 _______________________________________________ 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 _______________________________________________ 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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New Content on freeswitch.org
by Michael Collins 14 Jun '10

14 Jun '10

Hello there! Just an FYI, please visit http://www.freeswitch.org and check out the many new posts we have. Thanks! -Michael

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Re: [Freeswitch-users] Question about execute_on_answer
by Davide Colombo 14 Jun '10

14 Jun '10

Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" <msc(a)freeswitch.org> A: freeswitch-users(a)lists.freeswitch.org Inviato: Lunedì, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: <action application="export" data="nolocal:execute_on_answer=transfer myivrpresentation XML default"/> <action application="bridge" data="user/1019@${domain_name}"/> When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000(a)fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ 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 _______________________________________________ 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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