June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Want to play music on hold while executing a time consuming operation in Lua
by Sidharth Kshatriya 17 Jun '11

17 Jun '11

I am implementing an IVR using Lua in Freeswitch. In my Lua script I use curl to a web service. Sometimes the response takes a long time to come back. During that time period I would like to play music on hold. I have searched the freeswitch discussion archives a lot. There seem to be many suggested ways to implement music on hold from a Lua script but the answers are not very clear / not really applicable to my use case. I don't know what method I should use and which one is recommended. *Method 1:* Transfer the call to 9664 (music on hold extension). However the implementation for this solution for this does not seem to be available in Lua. For example: How would I transfer the call back? *Method 2:* Using bgapi uuid_park park the call and using uuid_broadcast play an audio file. Again what do I do to unpark the call..? Thanks, Sidharth [P.S. This is a repost, I apologize but I never got any responses on my previous email. Need help! :-) ] -- Sidharth Kshatriya www.sidk.info

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mod_spandsp event spandsp::rxfaxresult variables missing
by Weigel, Stefan 17 Jun '11

17 Jun '11

Hi list, yesterday I had a ODBC related problem. Therefore I used the latest snapshot. Now my scripts are not working properly (I'm doing an ESL connection and listening for spandsp::rxfaxresults events). With an older release (mid of may) I was getting lot's more info's (for example variable_current_application_data), now I'm only getting the information documented on the wiki page: fax-document-transferred-pages fax-document-total-pages fax-image-resolution fax-image-size fax-bad-rows fax-transfer-rate fax-result-code fax-result-text fax-ecm-used fax-local-station-id fax-remote-station-id What changed and how can I get back the needed information ? Thanks and best regards Stefan

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Message Numbers
by mitch Johnson 17 Jun '11

17 Jun '11

Can we have the messages seperated by their respective message numbers. I subscribe to quite a few lists and the FreeSWITCH list is the only one that doesn't transpose the message number down into the messages. It would make searching for specific messages much easier. Thanks, Mitch

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Kamailio Interop
by Spencer Thomason 17 Jun '11

17 Jun '11

Hello all, I've been working on upgrading our SBC setup which consists of FreeSWITCH instances between Kamailio 3.1 instances. In initial testing FreeSWITCH would fail on an ACK timeout. After looking at the traces, Kamailio was responding to the 200 with an ACK that has a branch=0 parameter. It is possible to disable this behavior by setting syn_branch=0 in kamailio.cfg which causes an MD5 hash to be used as the branch parameter. In researching this it seems that this has been around in SER for some time as the ACK following the 200 is sent statelessly regardless of a t_relay() function. See: http://www.mail-archive.com/users@openser.org/msg00697.html The question I have is there any way to get Freeswitch to work with an ACK with a branch=0? and if so should it? Gotta love SIP. :-) Spencer

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Unable to get TLS working
by Rob Hutton 17 Jun '11

17 Jun '11

I have been unable to get TLS working for several days despite beating my head against it, so I am sure I am doing something simple and stupid. Brian and Jeff have tried to help, but I haven't gotten anywhere. I would appreciate any help that I can get.... http://jira.freeswitch.org/browse/FS-3346 Thanks, Rob

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Making a external call to Mobile number
by ankIT WALiA 17 Jun '11

17 Jun '11

Dear all, I am trying to make an external call to a mobile number of India using the default ext number 1005. For doing this, I did the following things. First, I created a user at IPTEL.org, I added a SIP profile. I registered with IPTEL. I also added a dial plan regex for 91 - (10 digit number). But, because of some issue, I was not able to connect to IPTEL SIP. I was getting error on X-LITE about some server issue. then, I tried my VOIP account with ActionVoip which is a paid account and has SIP service. I could connect to my action voip account through X-LITE SIP. I made a SIP profile for Action Voip and registered the username with action voip in FS. On reload, it got registered using action voip SIP profile. I changed the dialplan to use actionvoip gateway. But still, when I am trying to call to my mobile number in India. The call is not able to connect. I think there is gap in my knowledge. Please help me I am new to FreeSwitch. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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Setting Up an Sip Trunk without Authentication
by Dave 17 Jun '11

17 Jun '11

Hi, I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address. I put <extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> in the default.xml under Dialplan When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above. I Appreciate any help you may offer. Dave Goodwin

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how to show unregistered users
by rentmycoder rentmycoder 17 Jun '11

17 Jun '11

Hi, It may be an easy question, but I'm unable to find the answer on wiki or google... sofia status profie internal show only registered internal extensions... how to list all users (both reged and unreged)? like: sip show peers in Asterisk... Thanks,,,

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mod_fifo shared lag
by Sascha Daniels 17 Jun '11

17 Jun '11

Hi together. Is there a possibility to have a shared lag over several fifos? In a callcenter with agents in several Queues the lag is 10sec. After a call from fifo A the agent should not get a call from fifo B within those 10sec. Basically like shared_lastcall=yes in Asterisk 1.6 Regards Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

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Test message - Please ignore
by Michael Collins 16 Jun '11

16 Jun '11

I <3 FS

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