- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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bypass_media problem
by David Villasmil 16 Jun '11

16 Jun '11

Hello Guys, I have a set where i'm receiving a call from a gw and sending to another gw. Both un-natted. If I set in my (lua) script bypass_media=false, call goes just fine, if i set it to true, FS sends "Temporary unavailable" to the B-side and CANCELs the A-side... Codec is fine as you will see in the pastebin.. here's the trace: http://pastebin.freeswitch.org/16480 and here's the console log: http://pastebin.freeswitch.org/16481 I'm bridging with a lua script, this is the pertinent part: gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no registration is needed.... session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407") session:execute("set","inbound-late-negotiation=true") session:execute("set","inbound-bypass-media=false") session:execute("set","proxy_media=false") session:execute("set","bypass_media=false") --session:execute("bridge","{sip_auth_username=" .. gw_sip_username .. ",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number .."@".. gw_sip_ip .."") fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" .. route_name .. ")}" .. out_number .. "") session:execute("bridge","{loop=3}sofia/gateway/${distributor(" .. route_name .. ")}".. out_number .."") -- hangup session:hangup(); Thanks guys for your help! David

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