July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Variable in Bridge string
by Ahmed Ajmal 15 Jul '10

15 Jul '10

Hi I have a strange problem creating a bridge string from LUA. The LUA script returns a bridge string which looks like this: sofia/gateway/${distributor(my_dist)}/1773XXXXXXX I pass this bridge string to the bridge app in the dialplan: <action application="bridge" data="${bridge_string}"/> On executing the dialplan I get the following error: Invalid data (${bridge_string} contains a variable I have tried executing the bridge app from within the lua script it works but I would prefer to have the dialplan process it. I am on Freeswitch 1.0.6. Thanks Ahmed

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Message Waiting Event
by Phillip Jones 14 Jul '10

14 Jul '10

Hi there, I am trying to see if I can create a SIP NOTIFY message with mesage waiting headers. I am using mod_managed and so far did the following in a little event loop: var mw_event = new FreeSWITCH.Native.Event("MESSAGE_WAITING", string.Empty); if (mw_event != null) { mw_event.AddHeader("MWI-Messages-Waiting", "yes"); mw_event.AddHeader("MWI-Message-Account", "sip:3475559995@64.40.xx.xxx"); mw_event.AddHeader("MWI-Voice-Message", "3/1 (1/1)"); mw_event.AddHeader("Profile", "internal"); mw_event.Fire(); } else { Log.WriteLine(LogLevel.Alert, "Failed to create message waiting event"); } However when at each AddHeader FS complians: "Trying to addHeader an event that does not exist!" mw_event is not null however. Can someone point me the right direction - on to create that event correctly. I tried SWITCH_EVENT_MESSAGE_WAITING and NOTIFYas the event type also. Thanks! Pj

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Conference events and mod_erlang_event
by David Swardstrom 14 Jul '10

14 Jul '10

I am unable to get any conference events with mod_erlang_event even when I use {event, 'ALL'}. I also tried {event, 'conference::maintenance'} and {event, 'conference', 'maintenance'} I also don't see any conference events showing up in the log file. I did get "PRESENCE_IN" events. Any suggestions? Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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Has anybody configured an Atcom AX-4S (BRI card) with freeswitch?
by Milena 14 Jul '10

14 Jul '10

Hello, I wanted to know if anybody has tried to install an Atcom AX-4S card (ISDN BRI) with freeswitch; If so, did it ever work? It is meant to work with asterisk using mISDN, anybody knows a workaround that might get it working in FS? (I'm willing to try crazy ideas but I need some pointers please) Any suggestions on an ISDN BRI card with 4 ports that has already been tested with freeswitch? (Sangoma AFT A500 only has 3) Thank you in advance. -Milena

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SIP2VXML
by Jan Berger 14 Jul '10

14 Jul '10

http://tools.ietf.org/html/rfc5552 Hi, the standard above describes a SIP 2 VXML interface. My question is can I do this with FreeSWITCH? Jan

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Bypass media after bridge
by David Aldworth 14 Jul '10

14 Jul '10

In order for bypass_media_after_bridge to work correctly what should the sip_profile settings be for media? Currently I have: <param name="inbound-bypass-media" value="false"/> <param name="inbound-proxy-media" value="false"/> Also, leading up to the bridge in the dialplan I have: <action application="set" data="bypass_media_after_bridge=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="ringback=$${us-ring}"/> FS does send a re-invite after the 200 Ok, but the SDP endpoints in the re-invite still reflect the local IP for media instead of bypassing. What to do? Much appreciated. David

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Active UDP connection to nameserver
by Puskás Zsolt 14 Jul '10

14 Jul '10

Hi. I just noticed today that one my FS server is keeping up an active udp connection to one of the configured nameserver in resolv.conf. Is it possible to disable this and connect when necessary ? Thank you for your help!

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Snom M3 and FreeSWITCH
by Desmond S. Fuller 14 Jul '10

14 Jul '10

Trying to get my M3 working with FreeSWITCH and not having any success. SIP Identity Status is giving me ERROR. I verified the settings with a softphone. SIP trace didn't give me much to work with. <code> Received from udp:74.107.124.42:5060 at 13/07/2010 22:36:21 (889 bytes) NOTIFY sip:1001@172.16.64.40;line=17173 SIP/2.0 Via: SIP/2.0/UDP 74.107.124.42;rport;branch=z9hG4bKDj6t71eeS6e7c Max-Forwards: 70 From: <sip:1001@74.107.124.42>;tag=57834eS1U31Sa To: <sip:1001@74.107.124.42>;tag=03pjgvabt070k Call-ID: ykp.do16d8bkr4v.dmey(a)74.107.124.42 CSeq: 133417359 NOTIFY Contact: <sip:1001@74.107.124.42:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:1001@74.107.124.42 Sent to udp:74.107.124.42:5060 at 13/07/2010 22:36:21 (641 bytes) SIP/2.0 200 OK Via: SIP/2.0/UDP 74.107.124.42;rport;branch=z9hG4bKDj6t71eeS6e7c Max-Forwards: 70 From: <sip:1001@74.107.124.42>;tag=57834eS1U31Sa To: <sip:1001@74.107.124.42>;tag=03pjgvabt070k Call-ID: ykp.do16d8bkr4v.dmey(a)74.107.124.42 CSeq: 133417359 NOTIFY Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include- Event: message-summary Subscription-State: active;expires=3600 Supported: timer, precondition, path, replaces User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 </code> Any ideas? - Desmond

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Sangoma A101 - 1 port E1/T1 Red Alarm
by ovvenkat 14 Jul '10

14 Jul '10

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Cannot register with flowroute.
by guru singh 14 Jul '10

14 Jul '10

Hi, I signed up with flowroute and was trying to setup an external gateway to use the test credits. I do manage to register using just a softphone without FS in the picture. I followed the provider example in the wiki. I have been given only the following SIP credentials from flowroute: SIP Registrar/Proxy username and auth-username: its the same value for both password I have iptables stopped and selinux disabled. I don't think its a NAT issue as I have an incoming DID from ipkall working perfectly fine. It's a 99% default config. I made no changes to either vars.xml or /conf/sip_profiles/external.xml Here is my /conf/sip_profiles/external/freeswitch.conf.xml http://pastebin.freeswitch.org/13426 SIP trace: http://pastebin.freeswitch.org/13427 Thanks gsin

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