- FreeSWITCH-users - FreeSWITCH Mailing Lists

Reason header in SIP 180 / 183
by jay binks 05 Jul '13

05 Jul '13

Hey all... I need to tap the collective knowledge of all. In the scenario where I can get a Reason header in a Progress response ( 180 / 183 ) how can I control how Freeswitch will respond. I can see 3 possible scenarios here, but it would be great to control which scenario was used. 1) Freeswitch just passes through the 180/183 with reason header... without doing anything with it. 2) Freeswitch can remove the reason header from the 180/183 and just play back the media stream that is accompanied. 3) Freeswitch can act upon the reason, and hangup the call .. ( if its a Q850 Cause code 1 for example ) now... where is this useful. well, what if I had a carrier that gave Q850 cause codes in ISDN progress messages ( number unallocated ) but then they played an RVA saying "number dialed is not connected". They have done it this way to allow customer to choose how to handle the RVA ( do you want the RVA or not ) but does FS give me the ability to truly control this ? ( without needing to go as far as complex ESL applications ) -- Sincerely Jay

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Is FreeSwitch run on Windows 2008 R2 64?
by Albert Nguyen 05 Jul '13

05 Jul '13

Hi, I am trying to load freeswitch on to Windows 2008 R2 64 bit server but so far I can not get any windows versions of freeswitch to work on this os (latest 32 bits and 64 bits version from freeswitch web site). It give error message saying that it could not load the ..v8.dll and another dll file within the switch_loadable_module.c. Is Freeswitch not compatible to win2008R2 or am I missing something? Cheers, Albert

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att_xfer question
by Stuart Mills 05 Jul '13

05 Jul '13

Hi all, I have a question which is hopefully not too complicated. I’m playing with mid call transfer features, I have an extension in features.conf to do the MCT which works fine, a_leg connects to the b_leg, then b_leg transfer to destination c_leg and can talk. My question is this, is it then easy for b_leg to toggle the call between a_leg and c_leg, I take it it’s just another feature extension that bridges calls between the 2 legs, could it be that simple? Kind Regards, Stuart Mills

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FFax T.38 how to force Freeswitch to reinvite with T.38
by Peter Steinbach 04 Jul '13

04 Jul '13

Hello, while receiving a fax and forwarding it to Hylafay via a tty modem interface, we currently only get T.30 analogous fax working. We have set <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> However Freeswitch does not offer a reinvite with T.38. The strange thing is: When we receive a fax with the rxfax application (and do not forward via tty modem to Hylafax), then Freswitch does offer a T.38 Reinvite and the fax is received via T.38. So <patton> ----> <freeswitch rxfax>: Freeswitch offers T.38 Reinvite to Patton <patton> ----> <freeswitch> --TTY modem--> <Hylafax>: Freeswitch does not offer Reinvite to Patton Anybody has a clue to overcome this? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Faxing problems if B leg offers T.38
by Rob Morin 04 Jul '13

04 Jul '13

Im having problems faxing if the B leg offers T.38. Several scenarios, but always the same result. First, the architecture FAX à ATA (T.38 optional) à FS à Carrier à Destination (Possibly another carrier (PSTN) and the fax machine) On my ATA, T.38 is enabled (Auto Detect). If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesnt support T.38, the B leg doesnt offer it and the faxes go through. Ive set Freeswitch as: <action application="set" data="fax_enable_t38=true"/> <action application="export" data="t38_passthru=true"/> When thats the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds. When I set Freeswitch as: <action application="set" data="fax_enable_t38=false"/> The B leg still offers T.38 if its other side is capable. Freeswitch doesnt pass the offer through and the B leg terminates the call after about 30 seconds. Im running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ). Ive had no success attempting to configure FS as a gateway. Frankly, T.38 passthrough has had the most success, mostly when the other ends arent T.38 capable and it stays in G711u. Is there something else I need to do to enable T.38 in passthrough mode? Thank you, Rob Morin PS I can provide tcpdumps of this, or whatever else is necessary.

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Caller ID and Caller Name Issue
by Jagadish Thoutam 04 Jul '13

04 Jul '13

HI, i have a freeswitch as a VOIP Gateway that will handle the Inbound and Outbound calls with media proxy, here issue comes My Freeswith is Sending FreeSwitch as a callid Number.. and callid name as a Number i wondered why it is sending like that i even comment the effective acller id name and caller id but still same issue, can any one help on this. Version : FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d 2013-04-23 19:49:07Z) Thanks Jagan

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how to configure pin for use in gsmopen ?
by Luis Azedo 04 Jul '13

04 Jul '13

Hi, how do i enter the sim card pin for use with Huawei K3715 ? thank you -- Luis Azedo

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SIP Trunking
by Ashwin Jain 04 Jul '13

04 Jul '13

Is there anyway in which we can authenticate SIP phones based on MAC addresses? -- Thanks and Regards, Ashwin Jain

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Upon re-INVITE, FS responds call does not exist. Help!
by Avi Marcus 04 Jul '13

04 Jul '13

PCAP of the SIP: http://ge.tt/923EDxk/v/0 I've got a lot of calls to a CLEC which the SIP says MetaSwitch. A about 930 seconds from the start, a timer kicks in with a re-INVITE but FS responds 481 Call Does Not Exist. Then the other side tears down the call, even though FS keeps sending media for another 10-246 seconds. pcapsipdump puts all the packets together, so why doesn't FS? The call ID seems to be the same. I'm on FreeSWITCH (Version 1.2.8 git 03f9daf 2013-04-24 20:52:10Z) I know it used to work with this carrier, and while they are investigating -- and claiming that it's not happening with anyone else -- I hoped someone could see something funny in the PCAP that would help. Thanks! -Avi

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attended transfer - fails
by Lloyd Aloysius 04 Jul '13

04 Jul '13

Hi All, I could not figure it out why the attended transfer is failing. UA1 - Call - UA2 UA2 Answer UA2 Transfer - Dial UA3 While UA3 is Ring UA2 Press the Transfer After transfer complete. *Issues* 1. Now UA1 hear the MOH only not the UA3 Ring back 2. After timeout the call never reach the UA3 voice mail ( This is the timeout option ) and the call is just hangup UA Bridge Data - {sip_invite_domain=alcan.mydomain.ca ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca ,originate_timeout=20}[leg_timeout=20]${sofia_contact( nelson(a)alcan.mydomain.ca)} Any help is appreciated. Thanks Lloyd

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