- FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL play_and_get_digits terminator problem
by Callum Guy 21 Jan '14

21 Jan '14

Good day All, I just wanted to report a minor event issue for application play_and_get_digits. Consider an application using a parameter string such as the following: 5 5 5 5000 # silence_stream://250 silence_stream://250 myvar \d+ 3000 In this scenario I would have expected the entry of # to terminate the application and produce a CHANNEL_EXECUTE_COMPLETE event however this does not occur. The actual result is that it terminates the current cycle, plays the invalid prompt and moves on to the next cycle until all 5 input cycles are complete. I was hoping to use the terminator to allow the application to exit. Perhaps this is the intended behaviour provided to allow a request such as the following to be terminated *within* the mix/max digit thresholds, so i will wait for confirmation. If this is the case then please let me know and i'll make it explicit on the wiki so this question isn't raised again. For now i have implemented a simple work-around where we allow 0 length input as follows and follow this with custom validation in my ESL handler: 0 5 5 5000 # silence_stream://250 silence_stream://250 myvar \d+ 3000 As an aside, I've tried my very best to find the DTMF queue processor in fisheye and haven't managed to find the method, can anyone point me towards the best resource for getting an understanding of the core code base? I'm sure that's a question that comes up plenty.. Best Regards, Callum

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Dynamic Xml (extension includes and gateway bridging)
by Yisroel M. Olewski 21 Jan '14

21 Jan '14

Hi About #1, I guess I'll have to change the xml or use curl (as ido suggested) About #2, can you show me a working gateway string example ? I understand I need to send proxy, username & password thanks You'll be better served if you change both your dialplan and your application to work in "outbound mode": http://wiki.freeswitch.org/wiki/Mod_event_socket#Outbound I don't think you can achieve #1 from within your application - at least not consistently - without adding a "park" entry to the dialplan. For #2, yes, it's possible. Just add the gateway info to the bridge string. I usually do something like: bridge sofia/internal/destination at GATEWAY_IP <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> , but you can use "sofia/gateway/destination" if you have the gateways defined on the XML files. From: freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [mailto:freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> ] On Behalf Of Yisroel M. Olewski Sent: Monday, January 20, 2014 5:55 AM To: freeswitch-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> Subject: [Freeswitch-users] Dynamic Xml (extension includes and gateway bridging) Hello everybody I have a .net application that connects to a FreeSWITCH instance and handles incoming calls I do not have easy access to the xml files I have 2 issues which I think might be similar 1. Parking: I need to preconfigure all incoming calls to park right away, otherwise the esl commands are ignored (as http://wiki.freeswitch.org/wiki/Mod_event_socket#Inbound ). How can I do this via code without needing to change the files physically? I haven't found any documentation for anything like this 2. Bridging: I need to be able to dynamically bridge to gateways listed in an external database, so it wouldn't be practical to keep updating the gateway files. Is it possible to bridge via esl to an arbitrary gateway with me providing host and credentials etc? Sincerely, Yisroel M. Olewski CEO [cid:image001.png at 01CF15D2.570A1DF0 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> ] +1 (718) 514-2721 +972 (52) 764-9971 Websites * CRM * ERP * Telephony * Inventory * Content Management Sincerely, Yisroel M. Olewski CEO +1 (718) 514-2721 +972 (52) 764-9971 Websites . CRM . ERP . Telephony . Inventory . Content Management

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Polycom phone not sending hold update when MOH is enabled
by Lou Allen Uysison 21 Jan '14

21 Jan '14

Hi, I have a Polycom SoundPoint IP 331. I am using uuid_phone_event <UUID> hold to put the phone on hold. If there is MOH to be played, the polycom only sends a media offer for the MOH instead of an actual HOLD signal. Due to this, Callstate does not change from ACTIVE -> HELD. Has anyone experienced this problem? Any way to make the HOLD signal still push thru with MOH?

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Bounty tasks around?
by Andrey Utkin 21 Jan '14

21 Jan '14

Hi, i'm freelance developer with experience in multimedia processing, and, some years before, VoIP. I have some contributions into FFmpeg, VLC and GStreamer, read more here http://blog.krieger.pp.ua/?p=1 . I wonder if i can help open source VoIP community on some multimedia related issues, including video processing (is VVoIP era already here, huh?). I have searched for similar bounty postings and haven't found a lot, but i decided to ask. I am ready to work on non-multimedia-related tasks, too. Of course it is desirable that proposed tasks are backed by reasonable payoff, although working on important FOSS issues is already a sort of award. Your suggestions to check other bounty-rich projects in aforementioned area are appreciated. Thanks in advance for any kind comments. -- Andrey Utkin

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mod_sms and "bridging" sms
by Donny Hardyanto 21 Jan '14

21 Jan '14

Hello everybody, I am trying to implement chatting using linphone and FS with mod_sms. I am confuse how to forward SMS from from a SIP gateway to FS SIP client (like bridge in sofia). I can confirm the SMS received from a external sip gateway to FS. Is there any "bridge" api for sms? Can anyone can point me the direction? Thanks Donny

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sending BYE due to ACK timeout
by Jason Sia 20 Jan '14

20 Jan '14

Hi, My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw. Thanks, Jason

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Freeswitch as outbound proxy server
by Sam 20 Jan '14

20 Jan '14

Hi Guys, I would like to know if we can use freeswitch as outbound proxy or not. If so then, can run script on outbound proxy fs server and validate and reject calls? If we reject the call from outbound proxy fs server then actual proxy/gateway call will automatically reject? Please let me know. Thanks in Advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-as-outbound-proxy-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Global Intercept
by Edenilson Perin 20 Jan '14

20 Jan '14

Even after that I answer an extension, if someone use the global intercept, he join the call. How can I make to intercept only when the extension is ringing? Regards,

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add dialstring to deflect
by Sayyed Mohammad Emami Razavi 20 Jan '14

20 Jan '14

Hello, How to set variables at deflecting?! any solution my technical friends?

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Dynamic Xml (extension includes and gateway bridging)
by Yisroel M. Olewski 20 Jan '14

20 Jan '14

Hello everybody I have a .net application that connects to a FreeSWITCH instance and handles incoming calls I do not have easy access to the xml files I have 2 issues which I think might be similar 1. Parking: I need to preconfigure all incoming calls to park right away, otherwise the esl commands are ignored (as http://wiki.freeswitch.org/wiki/Mod_event_socket#Inbound ). How can I do this via code without needing to change the files physically? I haven't found any documentation for anything like this 2. Bridging: I need to be able to dynamically bridge to gateways listed in an external database, so it wouldn't be practical to keep updating the gateway files. Is it possible to bridge via esl to an arbitrary gateway with me providing host and credentials etc? Sincerely, Yisroel M. Olewski CEO +1 (718) 514-2721 +972 (52) 764-9971 Websites . CRM . ERP . Telephony . Inventory . Content Management

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