July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS and SIP Proxy
by Adam Kelloway 15 Jul '11

15 Jul '11

Hi there, I have a FS installation that is on a private network, with a SIP proxy in front of it that has an assigned public IP. I would like to be able to make calls from a public network to this FS installation, but I am not able to fully establish SIP and RTP sessions. I have been playing around with the NAT features of FS without any luck. In this scenario, do you think it would be necessary for the SIP proxy to handle the NAT traversal to ensure that the SIP responses reach the SIP endpoint? For RTP, would the proxy need to modify the SDP response that FS creates, so that it includes the external IP address necessary to establish the session? What I am getting at is whether the proxy needs to be not a proxy at all, but more like some kind of SBC, that also has RTP proxy capabilities. It would be necessary to keep this proxy/SBC in the signal path for both sessions. Any guidance on this scenario would be appreciated. Thanks, Adam

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multiple active grammars supported?
by Yungwei Chen 15 Jul '11

15 Jul '11

Hi, I tried the pizza demo shipped with freeswitch, and I am wondering if multiple active grammars are supported. For example: question: was this the pizza you had in mind? response: yes/no (defined in grammar 1) Or help (defined in grammar 2; this case will be handled differently) Thanks.

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Invitation to connect on LinkedIn
by Steve Butterfield via LinkedIn 15 Jul '11

15 Jul '11

LinkedIn ------------ Steve Butterfield requested to add you as a connection on LinkedIn: ------------------------------------------ Zohair, I'd like to add you to my professional network on LinkedIn. - Steve Accept invitation from Steve Butterfield http://www.linkedin.com/e/xbphn8-gq59uq0k-1h/vPtmrrfmcGvVxWv84eLqdl0FlSkzdW… View invitation from Steve Butterfield http://www.linkedin.com/e/xbphn8-gq59uq0k-1h/vPtmrrfmcGvVxWv84eLqdl0FlSkzdW… ------------------------------------------ DID YOU KNOW you can conduct a more credible and powerful reference check using LinkedIn? Enter the company name and years of employment or the prospective employee to find their colleagues that are also in your network. This provides you with a more balanced set of feedback to evaluate that new hire. http://www.linkedin.com/e/xbphn8-gq59uq0k-1h/rsr/inv-27/ -- (c) 2011, LinkedIn Corporation

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Flex RTMP Client Added To Git Tree!
by Michael Collins 15 Jul '11

15 Jul '11

I thought many of you would be interested in this story about the RTMP Flex client: http://www.freeswitch.org/node/332 The sample Flex client has been added to tree, so feel free to start experimenting! -Michael

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No contact header on 302 message
by Vedran Zeljeznak 15 Jul '11

15 Jul '11

hello, i'm having trouble with 302 messages, it looks like FreeSwitch is not setting Contact header on Leg-A 302 message when 302 message occurs on Leg-B. There is a valid Contact header in 302 message on Leg-B. I've posted FreeSwitch console output on: http://pastebin.freeswitch.org/16806. Does anybody know how can I tell FreeSwitch to set Contact header on Leg-A 302 message from Contact header in 302 message from Leg-B? --- Vedran Zeljeznak

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rtmp_url folder
by Jack 15 Jul '11

15 Jul '11

Freeswitch.html has a spot to set the variable rtmp_url. What should be in the folder it points to? Thanks, Jack

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non-blocking bridge in Lua
by Mi Ke 14 Jul '11

14 Jul '11

Hi all ! I want to make a Lua app which will start with Leg A, collect input, initiate outbound Leg B and when it answers, periodically play decreasing series of beeps to Leg A depending on how many minutes left to disconnect (remaining time queried via odbc). My initial idea was to issue a non-blocking bridge command after Leg B session originate returns ring-ready and then to do the rest in while loop, but the bridge called by api:executeString("bgapi bridge legB_session legA_session") just returns Job UUID without actually bridging sessions. A usual freeswitch.bridge call blocks further script execution until unbridged, disallowing me to check Leg B states when it's active. Is there any elegant way to acheive my goals using Lua/Dialplan apps ? To create an app starting with Leg A and constantly watching for Leg B existense/states, playing corresponding feedback messages to Leg A ? Thanks in advance for yours hints/responses Mike

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please help!!! how to limit each FS registered user's active calls
by ran zhang 14 Jul '11

14 Jul '11

hi all: I'm trying to limit each registered user's active calls, so if that particular registered user pass the limit, FS stops sending any new incoming call to it and it can't make any new calls. I looked into 'limit' api, the counter increment everytime a call FS receives, regardless of which user it is for, and counter doesnt reset itself after the call is finished.

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(no subject)
by Pablo Silva 14 Jul '11

14 Jul '11

unsub

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mod_directory questions (prompt missing)
by Dave Horton 14 Jul '11

14 Jul '11

I'm trying to use mod_directory and after building and installing am having a couple of problems in my testing: First, it appears that there is a prompt missing: 2011-07-14 12:13:49.801293 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[directory/dir-press.wav] (en:en) 2011-07-14 12:13:49.801293 [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/directory/dir-press.wav] [System error : No such file or directory.] All of the other prompts appear to be there (under /sounds/en/us/callie/directory/) but this one is missing for some reason. I did a standard build from latest git as of a few weeks ago. Is this a known issue, and is there somewhere I could get this prompt? Secondly, right after I enter the DTMF digits for someone's last name it it reports that there is 1 match and spells out the name, etc, (and encounters the missing prompt) it then immediately tries to to a transfer to the extension that is currently active (which I named 'company_directory') rather than waiting for my DTMF input confirming that I want to be connected to the matching extension for the name I entered. See log snippet below. The way I have set this up for testing is that I added an option to the main ivr that does a menu-exec-app and transfers to the company_directory extension, which then executes the 'directory' app. I'm not sure why the directory app finished running before collecting my response (as to whether I wanted to search again or connect to the matching extension) nor why it then tried to do a transfer (again) to 'company_directory' and then hung up. 2011-07-14 12:13:48.513298 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[directory/dir-to_select_entry.wav] (en:en) 2011-07-14 12:13:48.533305 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16@8000hz 1 channels 20ms 2011-07-14 12:13:49.681295 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-07-14 12:13:49.921288 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2011-07-14 12:13:49.921288 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[directory/dir-no_more_results.wav] (en:en) 2011-07-14 12:13:49.941293 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16@8000hz 1 channels 20ms 2011-07-14 12:13:50.989241 [DEBUG] switch_ivr_play_say.c:1649 done playing file EXECUTE sofia/external/+15083084809(a)65.162.239.203 transfer(company_directory enum) 2011-07-14 12:13:51.109194 [DEBUG] switch_ivr.c:1622 (sofia/external/+15083084809(a)65.162.239.203) State Change CS_EXECUTE -> CS_ROUTING 2011-07-14 12:13:51.109194 [DEBUG] switch_core_session.c:1118 Send signal sofia/external/+15083084809(a)65.162.239.203 [BREAK] 2011-07-14 12:13:51.109194 [DEBUG] switch_core_session.c:711 Send signal sofia/external/+15083084809(a)65.162.239.203 [BREAK] 2011-07-14 12:13:51.109194 [NOTICE] switch_ivr.c:1628 Transfer sofia/external/+15083084809(a)65.162.239.203 to enum[company_directory@default]

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