January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Fix - IGNORE INFO DTMF(3) (This channel was not configured to use INFO DTMF!)
by IamTrying 22 Jan '14

22 Jan '14

I am using WebRTC of FreeSWITCH with latest codes to fix some issues. But i am stuck with this annoying WARNING showing: > 2014-01-22 10:18:04.869529 [WARNING] sofia.c:7746 IGNORE INFO DTMF(3) (This channel was not configured to use INFO DTMF!) I changed in the switch settings as mentioned in the doc: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF , but still i get this same error. <param name="dtmf-type" value="info"/> or <param name="dtmf-type" value="rfc2833"/> or <param name="dtmf-type" value="none"/> or <param name="liberal-dtmf" value="true"/> Can anyone kindly advise how to fix it please? Thank you Best regards IamTrying

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How to specify profile to the gateway, configured in user directory
by Dmitriy Shumaev 22 Jan '14

22 Jan '14

Hi, everybody. FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z). I have 2 interfaces in FreeSWITCH server: -eth0 with public IP, where all gateways are located. -eth1 with private IP, where all VoIP equipment (FXS, FXO GWs) are placed. The source [ http://wiki.freeswitch.org/wiki/Clarification:gateways ] tells about 3 different ways of configuring gateway. I use the 3rd one: freeswitch\conf\directory\default\<Gateway_name>.xml. But now several gateways are moved inside private network (with the help of VPN). So I need to configure these gateways to corespond with internal profile (now all packets to these gateways, for example, OPTIONS, go thru internal interface but with external IP as source IP). How can I do it for best? P.S. The 1st variant (moving gateway settings to /conf/sip_profiles/internal-eth1/) required in manual adding gateway's IP in ACL list since (as I see) there is no such a parameter as "cidr" in gateway tag (the next is *WRONG*:<gateway name="GW_Name" cidr="192.168.1.17">). But as I can see in this variant, gateway's variables do not set. What am I doing wrong? With best regards, Shumaev Dmitriy KBR Ltd.

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Help with advanced inbound number routing
by Nick M. Hill 21 Jan '14

21 Jan '14

________________________________ From: Nick M. Hill<mailto:Nick.Hill@bombardier-it.com> Sent: ‎18/‎01/‎2014 18:46 To: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: Help with advanced inbound number routing Hi all, We have freeswitch setup as a gateway to our lync server, however at the moment I can’t figure out how to force freeswitch to send a call from one trunk to a lync number and set the display name here is how it is currently working Number xxx-xxx-xxx comes in on trunk one --> freeswitch sends the call to lync/xxx-xxx-xxx --> Lync displays the DID freeswitch called. Here is the From field in the Invite Header FROM: "0114xxxxxxx"<sip:0781xxxxx@10.1.220.1>;tag=D5pv251yDNHNe TO: <sip:+44114xxxxxx@10.1.16.113:5068>;tag=f955586259;epid=07DCEB7C65 I need the call to somehow present the 0781xxxxx number in the To field as the display name is this possible?

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Gateway configuration with specified profile
by Dmitriy Shumaev 21 Jan '14

21 Jan '14

Hi, everybody. FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z). I have 2 interfaces in FreeSWITCH server: -eth0 with public IP, where all gateways are located. -eth1 with private IP, where all VoIP equipment (FXS, FXO GWs) are placed. The source [http://wiki.freeswitch.org/wiki/Clarification:gateways] tells about 3 different ways of configuring gateway. I use the 3rd one: freeswitch\conf\directory\default\<Gateway_name>.xml. For examle look at the attachment. I use it as all things corresponding with a gateway are at the same place: - ACL (<user id="ThruToInfoIndust_192.168.15.60" cidr="192.168.15.60/32">) - gateway (<param name="..." value="..."/>) - user variables ( <variable name="toll_allow" value="domestic,local"/> <variable name="IsFromInfoIndust" value="true" direction="inbound"/> ) P.S. This variant required only in <param name="apply-inbound-acl" value="domains-<profile>:public"/> in all profiles, where <list name="domains-<profile>" default="deny"> <node type="allow" domain="$${domain}"/> </list> P.P.S. Variable "IsFromInfoIndust" used as because of "apply-inbound-acl""domains-<profile>:PUBLIC" . Where is a transfer to default context in case of this variable in public context. But now several gateways are moved inside private network (with the help of VPN). So I need to configure these gateways to corespond with internal profile (now all packets to these gateways, for example, OPTIONS, go thru internal interface but with external IP as source IP). How can I do it for best? With best regards, Shumaev Dmitriy KBR Ltd.

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conference not working
by Joli Martinez 21 Jan '14

21 Jan '14

Hello, When an external user dials calls the main number they get asked to a conference number. They input it and get connected to a conference. When a local registered user dials the same conference number they get connected to a new conference. I need them both to be on the same conference. So if a user whether from internal or external dial a conference number they get transferred to the same conference. EX: External calls in and gets IVR. IVR says please enter conference number user inputs 3001 they go to 3001 Internal user dial from his phone 3001 he goes into conference 3001(a)X.XX.X where X is the IP address of the FS box. They don't see each other. Thanks,

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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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