June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] NAT problem: Use local IPv4 for specific LAN IP address range
by Ben 28 Jun '13

28 Jun '13

Hi All, Just to let you know (if anyone else has similar issues), I ended up created a virtual IP interface on the server, and created a new "LAN" profile that listened on this address. This allowed me to use the same port number. I then set a policy on the LAN profile to use the internal IP addresses for SIP/RTP. When a handset enters the internal LAN, I set up the firewall to re-write the DNS reply to the handset so that it uses the internal IP and not the public IP for the domain name it requests (can also be done with an internal DNS server). This was re-written to the virtual IP, which registers the handset to the LAN profile. This meant that the handset used internal IP addresses for SIP and RTP, instead of trying to hairpin the traffic out and then in on the public IP of the firewall. Cheers. From: Ben [mailto:ben122uk@gmail.com] Sent: 11 June 2013 10:57 To: 'FreeSWITCH-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] NAT problem: Use local IPv4 for specific LAN IP address range Steve, Sorry for the late reply (4 months!!!), the immediate need for a fix went away, until now. The problem with setting up an additional profile would be that the end user would have to change port numbers when moving from the internal network to an external network. Unfortunately the end user does not have access to these sorts of settings with the app we're using, and more importantly, I need this to be seamless between networks. Does anyone else have any ideas? I'm starting to think this problem is unsolvable with the current functionality. Implementing an equivalent of the Asterisk "nat.conf" with a "localnet=" line, which will force the Freeswitch local IPv4 address as the SDP for any requests originating from one of these "localnets" would hopefully resolve the issue. Ben From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 07 February 2013 13:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Use local IPv4 for specific LAN IP address range One simple option would be to have a 2nd profile listening on another port (eg 5080) that does not use ext-*-ip and connect to that internally. -Steve On 7 February 2013 11:44, Ben <ben122uk(a)gmail.com> wrote: Greetings all, A question on NATing in Freeswitch - I have my internal SIP profile to use a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch server is on a local LAN IP, for example 192.168.0.10. External SIP clients can access the server via the external IP, as there's a static NAT on the router to map the public IP to the private IP. I have another SIP client on a different LAN, but behind the same firewall, for example, 10.0.0.10. This client is allowed to talk to the server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to pass through. My challenge is, how do I get Freeswitch to recognise this internal SIP client as being allowed to use the internal IP for SIP/RTP communications, and therefore present the client with internal addresses in the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and ext-rtp-ip, Freeswitch tells the client to go via the public IP. The firewall has a policy to not accept traffic to the public IP if sourced from an internal device. I have explored the NATing options in Freeswitch, but can't find anything that fits my scenario. It seems to be more towards how to enable NAT in certain situations, not disable it. I have also looked at re-writing the SDP, but this re-writes the incoming SDP from the clients. My intention was to re-write the outgoing SDP from the Freeswitch to include the internal IP address as the RTP connect in address. I don't have much Asterisk experience, but I seem to remember a nat.conf file where you could specify subnets that were treated differently. These could be classed as internal subnets, and would be given the private IP in the SDP from asterisk, everything got the public IP. I'm also aware that I could create another SIP profile easily with different NATing/IP options, but I want to find out if there's anything I can do with the above first. Many thanks for the help! _________________________________________________________________________ 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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FreeSWITCH adds WebRTC support to new 1.4 BETA.
by Anthony Minessale 28 Jun '13

28 Jun '13

http://slashdot.org/submission/2750655/freeswitch-project-releases-version-… -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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Do not transcode when both sides offer g729
by Mino Haluz 28 Jun '13

28 Jun '13

Hi, Is there any way how to force FS do not use G729 licenses when both sides offer g729, so it can do passthrough? Mino

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what file to needed for tls SIP Trunk as client in Free switch
by Abdullah 28 Jun '13

28 Jun '13

hi, all hope you are all fine and good. i am doing setup for tls in my free switch server.i use different carrier for tls sip Trunk. i am little bit confused what file is need to connect other carrier . currently i requested to my carrier give my cert name CACAFILE.pem .but i am having error Incompatible Destination . please advice me , if i generate cert to my server (agent.pem and cacafile.pem ) and give them to my carrier or is this responsibility to carrier give me cert for its TLS sip Trunk. any idea , Many thank, -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/what-file-to-needed-for-tls-S… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Problem with freetdm+libpri (euroisdn): T316 timed out
by Peter Waldheim 28 Jun '13

28 Jun '13

Hi, I'm currently trying to set up a new freeswitch box and consistently run in a problem with pri-line initialization (I think). Basically I get "T316 timed out" for all but a very few channels, which get "Channel not opened, proceeding anyway" and become available. This does not seem to happen when using openzap instead of freetdm. The log file looks like this: 2013-06-27 08:25:17.560815 [WARNING] ftmod_libpri.c:1997 [s1c1][1:1] -- T316 timed out, resending RESTART request .. for all channels 2013-06-27 08:25:22.600817 [WARNING] ftdm_io.c:3013 [s1c1][1:1] Channel not opened, proceeding anyway 2013-06-27 08:25:47.560814 [WARNING] ftmod_libpri.c:1997 [s1c2][1:2] -- T316 timed out, resending RESTART request .. for all channels but the first 2013-06-27 08:26:22.600778 [WARNING] ftdm_io.c:3013 [s1c3][1:3] Channel not opened, proceeding anyway 2013-06-27 08:26:47.560815 [ERR] ftmod_libpri.c:1993 [s1c4][1:4] -- T316 timed out, channel reached restart attempt limit '3' and is suspended 2013-06-27 08:26:47.560815 [ERR] ftmod_libpri.c:1993 [s1c5][1:5] -- T316 timed out, channel reached restart attempt limit '3' and is suspended .. for all channels but the first three or so Eventually the first 3 or so channels are available. Any help with this problem would be greatly appreciated. Regards Peter

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mod_nibblebill errors [ERR] switch_core_sqldb.c:1126 SQL ERR:
by mrajib321＠yahoo.com 28 Jun '13

28 Jun '13

Hi, I have installed, compiled and loaded mod_nibblebill sucessfully but when I tried to test it with some data using MySQL I received the following error ============================================================ 2013-06-27 21:57:23.892791 [ERR] mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE id='2'] 2013-06-27 21:57:23.892791 [NOTICE] switch_ivr.c:1822 Transfer sofia/internal/1201(a)192.168.1.120 to XML[hangup@default] 2013-06-27 21:57:23.892791 [ERR] mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE id='2'] I have added two variables, "nibble_rate" and "nibble_account", to the user's directory entry for simple testing. User 1 <variable name="nibble_rate" value="0.10"/> <variable name="nibble_account" value="1"/> User 2 <variable name="nibble_rate" value="0.15"/> <variable name="nibble_account" value="2"/> ODBC connection to MySQL database works fine. I tested it using " isql -v freeswitch". I can connect and display the tables and all that. The settings in the nibblebill.conf.xml file are: <param name="odbc_dsn" value="odbc://freeswitch:freeswitch:freeswitchrw"/> <param name="db_table" value="accounts"/> <param name="db_column_cash" value="cash"/> <param name="db_column_account" value="id"/> Database name: freeswitch username: freeswitch pswd: freeswitchrw The database data is the following mysql> use freeswitch; Database changed mysql> select * from accounts; +----+----------------+------+ | id | name | cash | +----+----------------+------+ | 1 | x-lite-win | 20 | | 2 | linphone-linux | 30 | +----+----------------+------+ I am running freeswitch version 1.2.10 git 30be15f on a Linux Mint 14 machine. Any help as to what the error means or how to fix it would be appreciated. Thanks.

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Freeswitch as SIP conference server for Avaya?
by Bradley Mina 28 Jun '13

28 Jun '13

Does anyone have something like this in place? Beginning to learn about Avaya systems, however I'm much more familiar with FreeSWITCH and I was curious if there are any limitations that would prevent one from using FreeSWITCH as a conference server via SIP through an Avaya CM? Referencing the thread, "SIP Trunk to Avaya CM" I can see that you could register FreeSWITCH as a trunk, and route your calls to it but I was wondering if Avaya has preventative measures in place to urge customers to purchase some other product or license. Any help appreciated!

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Regular announcements while queued
by Brendan Ord 28 Jun '13

28 Jun '13

Hi there, I'm relatively new to FreeSwitch, however have some experience with Cisco call manager. We are looking to replace our Cisco UCM/UCCX gear with a freeswitch based callcentre. I am having difficulty with the best practice for a particular queue feature that we currently use - which is the ability to break out of the queue to either voicemail or back to the reception extension, the prompt for this plays every ~ 2 minutes. In freeswitch I've been using mod_callcenter which seems the best practice for this module is writing a script. Can a script be written that can use the ivr application, thus giving users the prompt+options on a regular announcement however without losing their position in the queue? The other option is mod_fifo, with the chime option that can be set. However this only really gives us a single option to break out of the queue (rather than opt 1 for voicemail and opt 0 back to reception). Wondering what advice a pro would give as to which queue module is best, and if this should be all contained in a script (that gets called by the dialplan of the queue). Many thanks, Brendan

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issue with quintum tenor and bridge_generate_comfort_noise
by Matthew Meek 27 Jun '13

27 Jun '13

We are migrating to using FS as an SBC between a telco and our Oracle Call Center Anywhere platform. Our current production calls are on a Quintum Tenor gateway using PRIs and we are moving to SIP via the SBC. The telco has a 5 minute RTP watchdog, and when Agents are working off-hook the RTP stream stops, requiring us to use bridge_generate_comfort_noise=true on the Agent leg. This works great as long as the inbound and outbound calls are both via SIP/Freeswitch. When the inbound call is via the Quintum Tenor (H323 to the CCA platform) and outbound is via the SIP/Freeswitch SBC, we get audio quality issues. The quality issue sounds like an overly-aggressive or poorly implemented VAD, where when the customer takes a small pause to talk, his or her voice is chopped for an instant when they begin speaking again. This is mostly apparent when the customer is reading off numbers... I do not understand why FS would be behave differently depending on the origin of the inbound call, as I am only adding bridge_generate_comfort_noise=true to the outbound call leg. Freeswitch is acting as a B2BUA SBC. I have packet captures on the outside interface of the SBC and can hear the audio issue in wireshark. Any thoughts? Thanks, Matthew Meek

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Odp: Re: can mod_httapi decide if the call should be answerred?
by Mariusz Czulada 27 Jun '13

27 Jun '13

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