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Attendant call transfer
by Suneel Papineni 03 Feb '10

03 Feb '10

Hi, I am trying to establish attendant call transfer using event sockets. 1. A call has come into Freeswitch from an external Gateway and this call is parked (it is configured to park all calls coming to freeswitch) {Caller A à FS} 2. Once the call is parked, I am sending a command to originate a call to another number connected to external gateway. {FS à Caller B}. Call is established between FS and caller B. ("api originate sofia/external/<another number>@<gateway IP address> 9999") 3. On receiving event message as "Application: Answer", I am sending another command to bridge call between A & B. ("api uuid_bridge <callerA UUID> <caller UUID>") 4. With this call is established between A & B, but there is a huge delay (appox 30 secs). I believe that FS is still in the call and might be this is creating delay (not sure). Could you please tell me if I am doing something wrong or process to achieve this scenario working. I tried in to transfer the call instead of bridging using the command ("uuid_transfer <callerA UUID> intercept: <caller UUID> inline"), but the response is same as above with huge delay. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* 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 Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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voicemail_greeting_number - question
by mbsip 02 Feb '10

02 Feb '10

Hi ALL, I am playing around with VM and want to play user recorded greeting instead of default one. I've scaned wiki Mod_Voicemail and found proper parameter "voicemail_greeting_number". Unfortunately there is a lack of example hence i dont know if it is already working. Aforementioned param was placed in /conf/directory/default/1000.xml file (param name="voicemail_greeting_number", i tried many values) The effect is that the default greeting is played. Is this param embeeded into FS right now? How to use it? Is there any other place I should do the changes? I am running FreeSWITCH Version 1.0.trunk (16456). Thx in advance. Maciej

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Sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101
by Jerry Richards 02 Feb '10

02 Feb '10

I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is endlessly logging the following errors: 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event lenght from boost rxlen=23 evsz=1031 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event lenght from boost rxlen=23 evsz=1031 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event lenght from boost rxlen=23 evsz=1031 Does anyone know what is causing this? I am using Wanpipe Driver wanpipe-3.5.8.6. I reinstalled the driver and rebuilt FS a couple of times. Thanks and Best Regards, Jerry

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Announcing... the FreeSWITCH Users Group! COME JOIN US!
by Darren Schreiber 02 Feb '10

02 Feb '10

Hey everyone! I'm writing to invite you all to sign up for a new set of events - FreeSWITCH Users Group Meetups. These in-person gatherings across the country exist to encourage you in creating awesome telephony and general communications software, hardware and services. Anyone interested in, working on or otherwise wanting to learn about FreeSWITCH and general telecommunications services should attend our meetings. Note that topics are not necessarily limited to just FreeSWITCH! Meetings will involve formal trainings, informal install fests, lots of Q&A time and general time to just share ideas and meet people. Each meeting has a formal topic to help kick things off, so don't worry if you are not sure what you can offer to the group - just showing up and asking questions is a great help! If you do have a specific topic you'd like to learn more about, email the group's organizer and we'll see what we can do. We'll kick off with three group locations across the country. You can sign-up to learn more about the scheduled meet-ups by using the links below: * San Francisco, California: http://www.meetup.com/fsusers/ * Manhattan, New York: http://www.meetup.com/fsusers-ny/ * Orlando, Florida: http://www.meetup.com/fsusers-orlando/ PLEASE SIGN-UP ONLINE SO WE CAN GAUGE INTEREST - You are just showing you might come - not committing to a date! Dates for each meet-up will be announced on the relevant websites in the coming months. We'll also have a VoIP dial-in for locations that will allow it. ---> If you'd like to start a meet-up in your area, please let me know. I'll help you arrange the meet-up spot, topics and advertising of the event. These events are currently free. The events are run by an independent FreeSWITCH enthusiast and are not associated with any corporation or formal group. Enjoy and we look forward to seeing you at our first meet-up! - Darren Schreiber the FreeSWITCH Users Group

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vm-disk-quota
by mbsip 02 Feb '10

02 Feb '10

Hi ALL, Maybe this question will be piece of cake for most of you, but it makes me think. I would like to configure "vm-disk-quota" for all users i have. I followed the wiki page and provided: <param name="vm-disk-quota" value="10"/> to /conf/directory/default/1000.xml After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail disk quota is exceeded" feedback No surprise for me because i had more less 10 voice mails already recorded (before the vm-disk-quota was set up). Strange is that increasing value even to 100 does not change anything. The same thing with deleting recordings from user directory. The only wayout is to set it to default value=0 (even FS shutdown doesn't change anything) I am wondering why vm-disk-quota produces "Voicemail disk quota is exceeded" all the time Where the module is looking for stored voicemail recordings. Below is part of my configuration. 1) /conf/autoload_configs/voicemail.conf.xml <param name="storage-dir" value="/usr/local/freeswitch/vm"/> 2) /conf/directory/default/1000.xml <param name="vm-disk-quota" value="100"/> 3) /vm/FS_ip_address/1000 is empty Thanks in advance. Maciej

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Sangoma A200 FXS callwaiting and dialplan statements
by Robert Hadley 02 Feb '10

02 Feb '10

I have a server that has Freeswitch (1.0.5.pre9) a Sangoma A101 (E1/T1) and Analog A200 cards using OpenZap. The cards are connected. The wanrouter version is 3.5.8.6 as the newer version didn't work for us yet. Also trying to test with the FS trunk but having other wanpipe driver issues. I have a question about using the A200 card. The intent is to hook up FAX machines to the two FXS ports. The desired behavior is the first call goes to the first FXS port one, and while port one is busy a second call goes to the second port. In the Freeswitch dialplan there is a statement to bridge to FXS port 1 and if that fails then bridge to FXS port 2. However, the first bridge never fails even when it's busy. I notice the A200 card supports call waiting. The FS debug messages show that is what is happening. Question: How do I disable call waiting on the A200 card when used with OpenZap? I am not sure about whether the hangup_after_bridge=true and continue_on_fail=true are necessary or in the correct spots. I've tried calling with these statements present or not and moved after the first bridge but doesn't appear to affect the status of the first bridge statement. Question: Are the hangup_after_bridge=true and continue_on_fail=true statements necessary and in the right spot? Dialplan statements: <extension name="fxsfax">  <condition field="destination_number" expression="^(101[0-1])$"> <action application="answer"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="openzap/FXS/1"/> <action application="bridge" data="openzap/FXS/2"/> Debug statements for second call to FXS/1 extension: (Notice that several of statements mention CALLWAITING.) 2010-02-02 14:00:55.193948 [DEBUG] switch_core_session.c:639 Send signal sofia/internal/1045@192.168.72.141:5060 [BREAK] 2010-02-02 14:00:55.193948 [NOTICE] mod_dptools.c:658 Channel [sofia/internal/1045@192.168.72.141:5060] has been answered 2010-02-02 14:00:55.193948 [DEBUG] switch_channel.c:182 sofia/internal/1045@192.168.72.141:5060 receive message [AUDIO_SYNC] EXECUTE sofia/internal/1045@192.168.72.141:5060 set(continue_on_fail=true) 2010-02-02 14:00:55.193948 [DEBUG] sofia.c:3787 Channel sofia/internal/1045@192.168.72.141:5060 entering state [completed][200] 2010-02-02 14:00:55.193948 [DEBUG] mod_dptools.c:768 sofia/internal/1045@192.168.72.141:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/1045@192.168.72.141:5060 bridge(openzap/FXS/1) 2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:1191 Connect outbound channel OpenZAP/2:1/ 2010-02-02 14:00:55.194948 [NOTICE] switch_channel.c:613 New Channel OpenZAP/2:1/ [bc08ce84-eff2-491c-b601-14b7329c71ea] 2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:1203 (OpenZAP/2:1/) State Change CS_NEW -> CS_INIT 2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/2:1/ [BREAK] 2010-02-02 14:00:55.194948 [DEBUG] ozmod_analog.c:78 Changing state on 2:1 from UP to CALLWAITING 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/2:1/) Running State Change CS_INIT 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/2:1/) State INIT 2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:390 (OpenZAP/2:1/) State Change CS_INIT -> CS_ROUTING 2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/2:1/ [BREAK] 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/2:1/) State INIT going to sleep 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/2:1/) Running State Change CS_ROUTING 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/2:1/) State ROUTING 2010-02-02 14:00:55.194948 [DEBUG] mod_openzap.c:413 OpenZAP/2:1/ CHANNEL ROUTING 2010-02-02 14:00:55.194948 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/2:1/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-02-02 14:00:55.194948 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/2:1/ [BREAK] 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/2:1/) State ROUTING going to sleep 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/2:1/) Running State Change CS_CONSUME_MEDIA 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/2:1/) State CONSUME_MEDIA 2010-02-02 14:00:55.194948 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/2:1/) State CONSUME_MEDIA going to sleep 2010-02-02 14:00:55.210047 [DEBUG] sofia.c:3787 Channel sofia/internal/1045@192.168.72.141:5060 entering state [ready][200] 2010-02-02 14:00:55.213006 [DEBUG] ozmod_analog.c:450 Executing state handler on 2:1 for CALLWAITING 2010-02-02 14:00:55.657988 [DEBUG] switch_rtp.c:2004 Correct ip/port confirmed. 2010-02-02 14:01:05.723334 [DEBUG] ozmod_analog.c:422 Changing state on 2:1 from CALLWAITING to UP Thanks, Robert

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Re: [Freeswitch-users] Sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101
by Jerry Richards 02 Feb '10

02 Feb '10

Yannick, That procedure had no effect on the problem. Any other ideas? Thanks and Best Regards, Jerry _____ From: Yannick Lam [mailto:yannick@sangoma.com] Sent: Tuesday, February 02, 2010 12:38 PM To: Jerry Richards Subject: Re: Sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 Hi Jerry, Can you please stop freeswitch, smg and the wanpipe driver. Then shutdown your computer and then re insert the card in the systema nd then boot again and see if you are still getting the issue. Thank-you, Yannick Lam Hang, B.Eng, Tech Support Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada e. yannick(a)sangoma.com | <skype:yannick.sangoma?add> Skype | <mailto:yannick_sangoma@hotmail.com> msn <mailbox://C%7C/Documents%20and%20Settings/Marc/Application%20Data/Thunderbi rd/Profiles/6u5xr3yx.default/Mail/Local%20Folders/www.sangoma.com> www.sangoma.com | <http://wiki.sangoma.com/> wiki.sangoma.com Lifetime Warranty. Because it must work! Jerry Richards wrote: I upgraded my FS version to 20100201 (i.e. Feb 1st 2010) and the console is endlessly logging the following errors: 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 2010-02-02 12:18:10.125121 [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event lenght from boost rxlen=23 evsz=1031 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 2010-02-02 12:18:12.134803 [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event lenght from boost rxlen=23 evsz=1031 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100 Expecting 101 2010-02-02 12:18:14.134440 [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event lenght from boost rxlen=23 evsz=1031 Does anyone know what is causing this? I am using Wanpipe Driver wanpipe-3.5.8.6. I reinstalled the driver and rebuilt FS a couple of times. Thanks and Best Regards, Jerry

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Subdomain directory
by Tim St. Pierre 02 Feb '10

02 Feb '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I'm building a multi-tenant system, and I need to separate customers from each other, but only in certain situations. I want to apply a subdomain to directory entries, so that a search for a user within a subdomain will only return a user in that domain, but a search in the parent domain will match all users in that domain AND all subdomains. Is there some sort of domain aliasing mechanism, or should I do this in the external scripting? I am doing everything with xml_curl. I could write the script so that it will look at the domain requested and decide if a user should be returned or not. Does anyone see any problems with this setup, or know of an easier way? The goal is so that things like SIP subscription and registrations are all done in the parent domain (only one domain system wide), but voice mail, directory application (dial by name), and some other features can be restricted to the same customer. All our user ID's (extension numbers for us) are unique. Thanks! - -Tim - -- Tim St. Pierre IP Voice technician Communicate Freely 1-877-291-8647 x5101 sip:5101@communicatefreely.net tim(a)communicatefreely.net -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.4 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQCVAwUBS2hvpoqVcvNCnHOrAQK9sgP/eiEtcEx1+OnMZPw0ZQ4gQUG+v/ddewa7 M8c5At4MoqjqNum5ruHtVA22UQt9U3gCG4gsXwaIXmDEDiMm6CTnjYYAgv5q4DvQ 0CmgHo/e1tgTly65GWIys2kwwUqyFsktZ0AeC7FFTyHLfr8l3uE4FJEY5lMRRdrC 0Eceg7MVoY8= =e45O -----END PGP SIGNATURE-----

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Fifo: ring agents without answering to caller
by Massimiliano Ravelli 02 Feb '10

02 Feb '10

Hi everybody. I am using asterisk for a call center pbx and evaluating if I can replace it with FS. With asterisk queue I can ring agents without actually answer to the caller so he won't pay till the agent pickup. Can I do the same with FS ? Moreover I'd really like to fullfill this scenario: I managed to solve it in asterisk only with an ugly workaround. Customer calls the queue and he doesn't get answered for 20 seconds and then he hears welcome message, music and whatever is configured in the fifo. Obviously the agents should ring even in the starting 20 seconds. Any hint or pointer ? Thanks in advance, Massimiliano

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Phones losing registration
by Dave Stevenson 02 Feb '10

02 Feb '10

Hi, I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSwitch. All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds. The Cisco and Thomson phones are OK, staying registered without problems but other phones, including SwissVoice IP10Ss and a cheap wireless VOIP phone lose registration after a random time, sometimes hours, sometimes days. Is this a common issue with known work-arounds or can someone point me in the right direction for how to see if any errors are being generated or if I need to turn on any additional FreeSWITCH logging options to trap the errors please ? regards Dave

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