January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Failed bridge in dial plan still answers an OpenZap inbound call
by Scott Ellis 14 Jan '09

14 Jan '09

I have an inbound call via OpenZap, when I attempt to bridge to a SIP extension, I get the ring tone (inbound line) up until the bridge fails (for timeout or do not disturb). At this point the call is answered and then my dial plan moves on to attempt another bridge to different extensions. So I no longer have the ring tone for the person dialing in. The call can still be answered and everything works ok, but I would rather not answer the call until someone actually picks up. Failing that simulating a ring tone would be good enough. Have searched around, but at a bit of a loss on how to dothis. Any suggestions greatly appreciated. Scott From my dialplan <extension name="LandLine IN"> <condition field="source" expression="mod_openzap"/> <condition field="caller_id_number" expression="^[1-8]$">   <action application="set" data="continue_on_fail=true"/>  <action application="bridge" data="{leg_timeout=30}sofia/$${domain}/500"/>   <action application="set" data="call_timeout=15"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="ring_ready"/> <action application="bridge" data="${group_call(ringgroup2@${domain_name})"/>  <action application="set" data="call_timeout=15"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${group_call(everyone@${domain_name})"/> <action application="hangup" data="NO_ANSWER"/> </condition> </extension>

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ilbc & alaw transcoding not working correctly?
by Hostinsky Miroslav 14 Jan '09

14 Jan '09

Hello all!, I am using freeswitch with Nokia E65 phone. When phone uses iLBC codec and I am trying "monkey demo" in the example IVR sound is choppy. When using alaw/ulaw there is no problem, it works OK. IVR prompts work OK with G711 and iLBC, there is no problem. So probably freeswitch is not able properly transcode between iLBC and G711 (when calling to external SIP peer?). Machine is powerful but it runs inside Xen (but load is minimal, I think this is not problem). So, what do you think? Where is problem? Thank you! -- Miro

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Registrar performance
by kokoska.rokoska 14 Jan '09

14 Jan '09

Hi all, could someone be so kind and share some data about concurrently registered users at FreeSWITCH from production usage? I have done few sipp testing and fall in doubt... BTW: I use xml_curl for directory and the data are served through: 1. apache, php, mysql 2. lighttpd, "c" binary, mysql and in both cases the performance is near the same - well, let say, not differs dramatically. When run kamailio with usrloc db mode 3 (no cache, all in db) on the same machine, performance is very good, so I hope the uderlaying system is setted-up properly. Thanks a lot! Best regards, kokoska.rokoska

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SVN Commits now tweet!
by Brian West 14 Jan '09

14 Jan '09

http://twitter.com/freeswitchsvn Check it out. ;) /b

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Internal vs. External Call, How to detect answered call?
by Klaus Teller 14 Jan '09

14 Jan '09

Hi, I'm unable to detect whether a call was answered or not. Using a custom Java API to connect to Freeswitch Socket Interface, i observed that: 1) session.originate("sofia/internal/1003%192.168.50.94") does block until the callee picks up while 2) session.originate("sofia/gateway/sip.gafachi.com/14156782222") does not block and returns +OK (and the channel identifier) even when the callee doesn't pick up. I was thinking that the channel Id is returned only when the callee picks up. Since this is apparently not the case, what is the definitive way to know if the remote user picked up? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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Announcing mod_easyroute
by Ken Rice 14 Jan '09

14 Jan '09

Greetings! I would like to announce the inclusion of mod_easyroute in FreeSwitch. mod_easyroute is a DID routing engine that uses an ODBC data source to figure out where to route calls. It uses a gateway concept and a number concept with a relation of one gateway to many numbers. Numbers are looked up in the database and a dial string and other useful information is returned into channel variables including a full dial string that can be passed to the bridge function for sending the call out. This module is not really targeted at the PBX user but at the ITSP community as an example of what can be done in C using the ODBC extensions built into the FreeSwitch core. See http://wiki.freeswitch.org/wiki/Mod_easyroute or http://www.freeswitch.org/node/158 for more information. Please note: if you are wanting to play with mod_easyroute, please ensure you have FreeSwitch compiled with ODBC support enabled (see http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core for ODBC setup information) If you are wondering why I would release such a thing, I have had this code sitting around for a bit now, a friend of mine needed the ability to route DIDs to different SIP based Application Servers for his business, so I wrote this for him, over the course of the past several months several people have approached me needing something similar and I re-used this code a few times for various consulting projects. A few weeks ago one of the guys in the IRC channel mentions he was about to write something like this for one of his clients, I mentioned I had this and he offered to "bounty" this functionality if I would release it. So here it is guys have at it. Thanks to the guys that sent me a few bucks for this code! Now to get a quick plug in for myself. I'm a consultant that works mainly with ITSPs and VoIP related projects, If you find yourself in need of some professional services to assist you in customizing or deploying FreeSwitch for your business, drop me an email off list and I will get back to you. With the growth that's happened with FreeSwitch up til now, I can't wait to see what 2009 will bring! Ken Rice Krice <at> rmktek <dot> com Ken4VoIP AIM

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NORMAL_TEMPORARY_FAILURE problem
by Alexandru Nedelcu 14 Jan '09

14 Jan '09

Hi, I want to know exactly what does this hangup_cause means: "NORMAL_TEMPORARY_FAILURE". I'm receiving lots of those. Is the SIP provider to blame, or is my setup? I took a look at the sip communication (Wireshark/tcpdump), and I couldn't find a response from the sip provider that matches. All I got was "487 Request Cancelled" and "486 Busy Here". Thanks,

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FreeSWITCH, MRCP and Perl
by Paul Herring 13 Jan '09

13 Jan '09

What would it take to put a budget together to for this project? Date: Tue, 13 Jan 2009 01:55:36 -0500 From: mszlazak(a)aol.com Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl To: freeswitch-users(a)lists.freeswitch.org Message-ID: <8CB436312A08329-80C-1B5D(a)MBLK-M24.sysops.aol.com> Content-Type: text/plain; charset="us-ascii" "My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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FreeSWITCH, MRCP and Perl
by David Knell 13 Jan '09

13 Jan '09

Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

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Gateways vs Directory Users
by Apostolos Pantsiopoulos 13 Jan '09

13 Jan '09

I think one of the most difficult ideas to grasp when someone migrates to FS from other soft-switches is the definition of endpoints (in the generic meaning of the word) i.e. other gateways that your FS gateway will exchange traffic with. From what I understood from the wiki : a user (directory) is an entitiy that defines our subscribers (with registration or not) a gateway is another VoIP gateway that we use to send traffic to (provider) and sometimes receive traffic from (correct me If I am wrong this far) Now, I have the following scenario : I have an FS box that will exhchange traffic with certain gateways. Sometimes a gateway will act as a client (customer) or a provider, depending on the traffic direction. 1st approach : The first plan I tried, was to use the directory to "define" these other gateways since they are clients. I turned off registrations in my SIP profile (because I don't need any), I wrote an xml file for each gateway in the directory with the appropriate cidr and that was it. I then realized that I could not use my definitions in the users directory so that I can send a call to that client. In order to do so, I had to define that particular IP as a gateway in my SIP profile <gateways> section. 2nd approach : I forgot all about the directory thing (after all I needed not any mailboxes or registrations) and tried to implement the same scenario by just using gateways. I declared the gateways in my SIP profile and was able to dial out to them just by using : <action application="bridge" data="sofia/gateway/<gw_name>/$1"/> which was fine. But when I got an inbound call from that gateway I realized (1) that I had to define this gateway in an acl and (2) the dialplan used (context) was always the inbound one since there is no way of telling a gateway which context to use when accepting a call. I found a way in the wiki that suggests using DIDs to accomplish that, but I find it this to be not a very elegant solution because I don't want to use prefixes or DIDs to accomplish my goal. I want to be able to define a gateway using a gateway name (label) of my choice and an IP address, so when I come to the point of having 100+ gateways i won't have to search wich DID to use for a new one. 3rd approach : Use a combination of the above. Declare a gateway that receives and sends traffic twice. Once as a user in the directory and once in the gateways section of my profile. Does all these sound correct to you? Am I missing something? Which approach is the best? Please send me your feedback. Inline comments would be ideal. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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