January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_skypiax - help needed with modified snd-dummy (one way audio)
by Peter P GMX 09 Feb '10

09 Feb '10

I have mod_skypiax working nicely so far with 2 Skype channels. Thanks to all contributors, excellent work! In order to have more than 8 channels working, I have followed the instructions in http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System ist Debian 5.0R3) It compiled well however when I start snd-dummy I only have one-way-audio and my logs show Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver, /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920 working on a machine with 250HZ kernel Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using obsolete setsockopt SO_BSDCOMPAT Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages suppressed Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using obsolete setsockopt SO_BSDCOMPAT Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages suppressed If I reinstall alsa from deb everything sworks fine again (of course with the current limitations). First question: Has anybody had this issue before? How can I solve this? Second question: As I do not need 64 channels or more: how do I manage, that Skype instances 9..15 use a second instance of snd-dummy as addressed in the wiki? Best regards Peter

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Trunk Version Number
by Dave Stevenson 08 Feb '10

08 Feb '10

Hi, Running the latest SVN (16453) under Windows, the console "Version" command displays :- "FreeSWITCH Version 1.0.trunk (UNKNOWN)" Should the version number not include a meaningful build version in the brackets ? regards Dave

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Event socket: filter delete isn't working
by Nagalenoj H. 04 Feb '10

04 Feb '10

Dear friends, I've tried to delete the filter which I applied for an unique id. But, it doesn't work. After executing 'filter delete', I am receiving the events from that uuid. I used the command as 'filter delete unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'. I did the following operations. Made call to the event socket. Registered events for all. (events plain all). Applied filter for the uuid. (filter unique-id aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565). I've got a new uuid by using create_uuid. Applied filter for this new uuid. (filter unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565) Originated a call with that uuid. Now, I could receive events from both uuids. (Tested by giving DTMFs in both end and checked unique-id in event header). Then, I wanted to delete a uuid from the filter. (filter delete unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565). I thought, i won't receive the events from this deleted unique-id. But, I received the dtmfs from both unique-id. I'm using 'FreeSWITCH Version 1.0.trunk (15982)'. -- Regards, Nagalenoj H.

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Has anyone had Freeswitch crash on loading mod_voicemail?
by Mark Sobkow 03 Feb '10

03 Feb '10

I tried a "load mod_voicemail" in fs_cli, hoping to see what configuration section it requested from Erlang, but instead of loading the module, Freeswitch crashed without any error messages. SVN 15188 built on Ubuntu Hardy 32-bit.

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Need Help to setup freeswitch with sangoma card
by Thangappan.M 03 Feb '10

03 Feb '10

Dear all, I have successfully configured wanpipe with freeswitch. When I was the running wancfg_fs script the following files openzap.conf , autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf are created. I started the wanrouter command then executed the freeswitch. When I was executing freeswitch mod_openzap.c said the error as "Error for finding the span id. name:PRI_1". But in the openzap.conf and openzap.conf.xml files the span name is smg_prid. Why the freeswitch is referring the span name as PRI_1 ? Whether this has to configured in anywhere? In the freeswitch CLI using oz command I tried to dump the PRI_1 span id but it said te error as "PRI_1 is not found". When I was trying the command 'oz dump smg_prid' all the channel states and details shown. It seems that smg_prid span configured in openzap perfectly (Its my assumption). Then Why freeswitch is referring the span name as PRI_1. DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO? Could anyone please help me? REFERENCE: openzap.conf [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <boost_spans> <span id="smg_prid">  <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> </boost_spans> <analog_spans> </analog_spans> </configuration> -- Regards, Thangappan.M

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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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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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409 Conflict When registering FS with Softswitch
by RR 02 Feb '10

02 Feb '10

Gentlemen, I have a probably a simple problem but I have no idea why it's occurring. I am beyond novice/new to configuring FS and as my first try, tried to configure it to register with our softswitch so I could have the basic registration to happen between FS and it. However, I keep getting the "409 Conflict" message when that username is not actually registered with the softswitch. The profile / config file under sip_profiles/external is configured like so <include> <gateway name="Test-Inbound"> <param name="username" value="user1"/>--> <param name="password" value="user1pwd"/>--> <param name="realm" value="mysoftswitch.net.au"/>--> <param name="extension" value="1000"/> <param name="from-user" value="user1"/>--> <param name="expire-seconds" value="60"/>--> <param name="register" value="true"/>--> </gateway>--> </include> Any ideas why? And how to resolve it or what kind of traces I can send to you guys to help me resolve it? Like I said I am so new to FS that I don't even know what I can turn on in FS to see what's going on Thanks a lot \RR

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Somebody help me understand the 'features' set up please :-)
by mailinglist 02 Feb '10

02 Feb '10

Ok, I've gotten the Freeswitch to register to my VoIP provider. I've gotten my phones to register to Freeswitch, and I can receive and make calls, all very nice. I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the Freeswitch. When I receive a call, I would like to be able to transfer the call to another phone, or change the call to a conference call with two local phones. So I've been looking at the examples in the wiki, and I can't make them work, not as I understand them anyways. Especially the att_xfer seems to be able to do what I need. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer As I understand Example1, I should answer the call, and then press *3 during the call, and either transfer it or change it to a threeway call. I get the first part, create an extension in the dialplan called att_xfer. But what is meant by the second par 'then bind this feature to DTMF 3', how do I enter that, and where? I hope somebody can help me with this (again)? Best regards Fribse /Running Freeswitch on pfSense running on a VMWare ESXi 4.0

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Hybrid Encryption?
by Nicholas Lee 02 Feb '10

02 Feb '10

The TLS wiki page talks about [1] Freeswitch being able to act as a past though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a SIPS+SRTP session. Is there howto guide for the above? I'm also wondering if Freeswitch could do a drop-in encryption. ie. Phone <-> Asterisk replaced with Phone <-> FS <==> FS <-> Asterisk, where <==> is encrypted. Nicholas [1] http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption

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