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Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20)
by Anton Kvashenkin 27 Mar '12

27 Mar '12

Hello guys. I have setup the gateway for outbound calls with credentials provided by my ITSP. http://pastebin.freeswitch.org/18749 Trying to make test call using, for example, originate command originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif) result: ITSP proxy sends 500 Gateway is Invalid http://pastebin.freeswitch.org/18750 But, trying to make call using asterisk with the same credentials suprisenly works. Here dump of success call http://pastebin.freeswitch.org/18751 I also attach logs that provided mys ITSP. Don't mind that there is ip address from privite LAN, the first try was from behind NAT. The second, that i provided above (http://pastebin.freeswitch.org/18750) was from vps with staticly assigned address. For the first glance I thought that the root of this problem was NAT but IMO it's not. Also, my external.xml http://pastebin.freeswitch.org/18752

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Re: [Freeswitch-users] Binary download with curl
by Peter Olsson 27 Mar '12

27 Mar '12

If you follow the instructions on the wiki the build should work out of the box. The most common problem is that you've enabled autocrlf in git, which causes the build to break. If you still have problems after following the instructions, please report to Jira, since this is something that should be working. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Olli Aro Skickat: den 26 mars 2012 10:05 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Binary download with curl Hi all, Been trying to compile Freeswitch on Windows in order to get in the XML curl module, however it is not looking easy (76 more fatal errors to figure out...) Before I start that task I thought it is always worth asking :) Would anyone happen to have a download link to already compiled Windows 32-bit binary for close to recent version of Freeswitch with the XML curl module? Regards, Olli !DSPAM:4f707e2932761654319646!

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Cli command
by Luis Daniel Lucio Quiroz 27 Mar '12

27 Mar '12

Just wondering where i can read about cli commands LD

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Trouble with transfer and voicemail password
by Trever L. Adams 27 Mar '12

27 Mar '12

Hello everyone, I hope someone can help me. I am using my home as a setup to test FreeSWITCH. I am using FXO for my incoming and FXS to phones around the home. I am trying to do blind transfers when *0# is dialed on the internal phones for calls coming from outside the home. I currently have: <extension name="setup bind_digit_action_transfer"> <condition field="destination_number" expression="^SETUP_TRANSFER$"> <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/> <action application="bind_digit_action" data="do_transfer,*0#,exec:execute_extension,DO_TRANSFER XML Incoming-FXO"/> <action application="digit_action_set_realm" data="do_transfer"/> </condition> </extension> <extension name="setup transfer" continue="true"> <condition field="chan_name" expression="^FreeTDM/3:1/"> <action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/> <action application="set" data="bridge_pre_execute_bleg_data=SETUP_TRANSFER XML Incoming-FXO"/> </condition> </extension> <extension name="bind_digit_action Do Transfer"> <condition field="destination_number" expression="^DO_TRANSFER$"> <action application="log" data="INFO Transferring external call to another internal location..."/> <action application="read" data="0 10 NULL int_extension 10000 #"/> <action application="transfer" data="-bleg ${int_extension}"/> </condition> </extension> The transfer immediately starts ringing the entire house (1000). This is true even when I am trying to get it to transfer to 1003. Also, http://jira.freeswitch.org/browse/OPENZAP-173 seems to be back in that when the incoming phone hangs up, it seems to continue to ring. Maybe I have this setup all wrong. I do not want anyone outside (the call originator) to be able to initiate the transfer. I want the transferring party to be cut out of the call. I want the outside and the *0#INTERNAL_EXT# to be connected. As for voicemail, I all FXS, no SIP for now. How do I set the password? How would I setup a common voicemail box that every thing goes to except one extension? That should have a separate box and password. Thank you for any help. Trever -- I love dogs, but I hate Chihuahuas. A Chihuahua isn't a dog. It's a rat with a thyroid problem. -- Unknown

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Fax on Sangoma
by Anita Hall 26 Mar '12

26 Mar '12

Hi My client had been using an old torrenta cards which game problems with pretty much everything :) After evangelizing the cause of Sangoma for more than 3 months (yes, they are dumb), they are finally making the transition so I am pretty excited. The choice of card is A108D which does a whole bunch of DSP in the hardware. Now, the first hurdle that I have to make Sangoma jump across is getting incoming Fax over E1 right. We are using mod_spandsp, of course. So, here I will be needing a whole lot of help from the veterans of spandsp and Faxing :) I desperately need Sangoma to pass the Fax test or they will give me torrenta cards all over again! The primary cause of failures are - (49) The call dropped prematurely and (48) Disconnected after permitted retries. For example, in this case, can I conclude that the other end did not provide a Fax tone or is it something else? 2aeb5f7c-75b5-11e1-8f36-b3286880c45b EXECUTE FreeTDM/4:2/47615728 rxfax(/srv/fax/in/2aeb5f7c-75b5-11e1-8f36-b3286880c45b.tiff) 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] switch_core_codec.c:216 FreeTDM/4:2/47615728 Push codec L16:70 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] mod_spandsp_fax.c:1373 Raw write codec activation Success L16 2012-03-24 18:57:57.844876 [DEBUG] ftmod_wanpipe.c:965 [s4c2][4:2] First packet read stats: Rx queue len: 1, Rx queue size: 10 2012-03-24 18:57:57.904879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:57.924878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:58.104877 [DEBUG] ftmod_wanpipe.c:901 [s4c2][4:2] First packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 30 2012-03-24 18:57:58.444908 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:58.644907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:58.664878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:58.784878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:58.804879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:59.244886 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:59.464887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry number 1 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx type 0 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx type 4 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Sending ident 'Sangoma Fax Ident' 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: CSI without final frame tag 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: ff 03 40 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 4:1 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] Changed state from UP to TERMINATING 2012-03-24 18:58:00.524877 [DEBUG] ftdm_state.c:511 [s4c1][4:1] Executing state processor for TERMINATING I will need some more hand-holding with logs later :) regards, Anita

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no DTMF detection over VoIP
by Neil Patel 26 Mar '12

26 Mar '12

Hi All, I have a basic IVR application in Lua connected to a PRI line. It currently is not responding to DTMF input given from any VoIP call (e.g. Skype). However, it accepts input from local mobile or landline calls just fine. I recently pulled latest from git and built; before that, DTMF detection from both were working. Is there something I need to configure to allow DTMF for VoIP? Thanks, Neil

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acl.conf.xml don't understand
by Bzzz 26 Mar '12

26 Mar '12

Hi list, There's something I don't understand: if I add my LAN cidr (allow) in the second block of acl.conf.xml (domains), I can't call anymore. <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/>  <node type="allow" cidr="192.168.1.0/24"/> </list> From what I understood of the wiki, this should work because the block denies everyone but my line allows for LAN cidr. I guess I misunderstood something. JY --

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Language Files
by Volker Wend 26 Mar '12

26 Mar '12

Hi, where can I find german language soundfiles fort the voicemail module ? Br Volker

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VoiceChanger for FS
by Yuriy Nasida 26 Mar '12

26 Mar '12

Hello guys, If anybody know any similar implementation of VoiceChanger (or something like this) for FS ? Thanks.

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FLEX Client Authentication
by Nandy Dagondon 26 Mar '12

26 Mar '12

hello, how can i authenticate Flex client? i have tried the following in the user directory: 1. plain text password => (authentication failed) 2. a1-hash w/ using ssl md5 generated "username:domain:password" value => (empty password not allowed) 3. md5 generated plain text password => (authentication failed) peeking at rtmp.c, i can see it looks for the "password" parameter. so, how to generate the authentication values? i appreciate for your help. tks Anthony for the Flex client. it's cool! -nandy

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