- FreeSWITCH-users - FreeSWITCH Mailing Lists

incoming call stop working after a few minutes
by Philippe Le Toquin 08 Apr '11

08 Apr '11

Hi, I have the latest FS installed on my Guruplug and it was working fine overall Last Friday I decided to rename the hostname of the guruplug and since then I have problem with incoming calls no longer going through. I can't see how the 2 can related but that is the only thing I can think of. Also if I make a outgoing call then incoming will start working again for a few minutes and stops again The only thing I can see in the debug is that message: 2011-04-03 12:39:00.820680 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [8926(a)172.21.0.20] from ip 172.21.0.50 But I was seeing it before as well so it can't be only that. Is there some other trace I can activate to see more? Regards /Philippe

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Setting SIP request URI in bridge
by Pete Kelly 08 Apr '11

08 Apr '11

Hi I need to bridge a call to an IP a.b.c.d, however the request URI needs to be sent to a domain rather than an IP e.g. the UDP packet is sent to a.b.c.d, but the INVITE looks like "INVITE sip:12345@my.domain.com" This is the syntax I am using for the bridge: <action application="bridge" data="{sip_invite_req_uri= sip:12345@my.domain.com,sip_from_uri=sip:$1@my.domain.com ,codec_string='G729,PCMU,PCMA'}sofia/mediaserver/12345(a)a.b.c.d"/> however freeswitch just 503's the bridge with "NORMAL_TEMPORARY_FAILURE". If I remove the sip_invite_req_uri part, the call is bridged fine, but the INVITE line is incorrect. Is it possible to do what I need in Freeswitch? Thanks Pete

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how to send a "uuid_displace" command in C++ code?
by Frankie Yiu 08 Apr '11

08 Apr '11

Hi there, I would like to know how I can send a "uuid_displace" command in my c++ code. In C#, I can call Api.ExecuteString("uuid_displace " + <uuid string> + " start " + <strFileName> + " 20"); but in C++ how can I send this command? Hope someone can help me. Thanks in advance!! Frankie

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How to replay an audio when a ' * " (star key) is pressed?
by Frankie Yiu 08 Apr '11

08 Apr '11

Hi there, I would like to play an audio to the channel and then if the callee presses a ' * ' and the audio would replay immediately from the beginning, how can I do that? I am using C#. Currently I am calling PlayAndGetDigits() and would play an audio. I am also subscribing a DTMF event, when I find a DTMF = ' * ' I would call this command "uuid_displace " + <unique ID> + " start " + <file name> + " 20" " to play the audio again. The problem is that it seems like the event come in after 6 or 7 sec after the callee presses the key. What am I doing wrong or is there a better way to do this? Thank you. Frankie

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FS -How to limit the max number of registration users
by fieldpeak 08 Apr '11

08 Apr '11

Could anyone help advise how to limit the max number of registration users on FS? Thanks if any advice. Regards, Charles

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Transfer attempt for a previously a replaced call fails
by Joegen E. Baclor 08 Apr '11

08 Apr '11

Hi List, I have a scenario where a bridged call has been replaced due to a consultative transfer. This works pretty well and audio is bidirectional. I have the original uuid of the call in a var somewhere. The trouble begins when I uuid_deflect the bridged call once again to attempt another transfer. Sofia disconnects the channel. I am using the original uuid of the call (uuid prior to replaces). Is this the right way of doing it? Joegen

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Freeswitch
by Kaushal Shriyan 07 Apr '11

07 Apr '11

Hi, I have couple of questions regarding Asterisk. a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone numbers? b) Does it Support VoiceXML ? c) What PRI Card is recommended for using Asterisk ? Thanks Kaushal

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xml_curl response for voicemail_inject
by Frank Park 07 Apr '11

07 Apr '11

I had a quick question. I am trying to figure out what directory response is expected when a user wants to forward a voicemail to another extension. Looking at the post request of xml_curl, I see that it's invoking voicemail_inject. I've tried the same response as other voicemail function, which is similar to the authorization response, but that didn't seem to do it. Anybody care to share an example response when voicemail_inject is requested? Thanks, Frank -- ----=======================---- Frank Park Telonium Communications, LLC frank(a)telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----

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Job Opening at Barracuda Networks / CudaTel - Technical Escalation Engineer
by Anthony Minessale 07 Apr '11

07 Apr '11

Looking for someone well-versed in debugging voice and data using common tools like wireshark etc to take escalation incidents from tech support at CudaTel. Should be well versed in unix systems and internet tools and above average diagnostic skills. Preferred location or relocation to Ann Arbor MI but will entertain applicants from the bay area in CA as well. Please reply with resume to jobs(a)freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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FS does not relay BYE
by Eric Beard 07 Apr '11

07 Apr '11

Hello, I have just started using freeSwitch as a way to terminate calls from Microsoft Speech Server to voip gateways. I have almost everything working with a few exceptions. One of the problems I am having is that the final BYE issued by the terminator does not get relayed back to MSS, so MSS keeps the call open for an additional minute, then issues its own BYE, which freeSwitch can't match up to a call because it tore the call down already. The sequence: - MSS running on my machine originates a call, sends INVITE to freeSwitch running on a separate machine, with an internal and external NIC. - freeSwitch relays the INVITE to the gateway (in this case Affinity, but I get the same behavior with other gateways). - My cell phone rings, I pick it up, then hang up the call. - The gateway issues a BYE to freeSwitch, freeSwitch says OK and tears down the call without passing on the BYE. If I originate a call from my machine with a soft phone, it works fine. The only difference I can see is that the soft phone uses UDP, while MSS only talks SIP over TCP. I have pasted logs for the session at http://pastebin.freeswitch.org/16037. Thanks! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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