- FreeSWITCH-users - FreeSWITCH Mailing Lists

T.38 gateway and peer/self
by Peter Steinbach 24 Feb '17

24 Feb '17

Hello, we have the following screnario * The caller, a local fax machine (Fax, ATA and Freeswitch) with T.38, is behind a slow line so therefore T.38 is needed here * a central Freeswitch fax server serves as a central gateway to the SIP provider * the SIP provider sometimes reinvites T.38 and sometimes not. * In case, the SIP provider does NOT answer with T.38, the central Freeswitch fax server shall act as a T.38 gateway and transcode between local fax (T.38) and SIP provider (T.30) * In case, the SIP provider does answer with T.38, the central Freeswitch fax server shall pass T.38 to the local fax ,machine (T.38) How we set this up * according to the wiki we have a dialplan as follows: (this is what CLI shows) Dialplan: sofia/internalvpn2/49274xxxxx(a)10.7.0.1 Action set(fax_enable_t38=true) Dialplan: sofia/internalvpn2/49274xxxxx(a)10.7.0.1 Action set(fax_enable_t38_request=true) Dialplan: sofia/internalvpn2/49274xxxxx(a)10.7.0.1 Action set(execute_on_answer=t38_gateway self) Dialplan: sofia/internalvpn2/49274xxxxx(a)10.7.0.1 Action ring_ready() Dialplan: sofia/internalvpn2/49274xxxxx(a)10.7.0.1 Action bridge(sofia/gateway/sip.provider.com/xxxxxxxxxxxxxxx@sip.provider.com) In all profiles, we have <param name="t38-passthru" value="true"/> In case, the SIP provider does NOT answer with T.38, it works. The caller's side is reinvited with T.38, the SIP provider side is T.30 What the problem is: When the SIP provider' side IS answering with T.38, then (sometimes, when I send it via cli from the local freeswitch near the sending fax machine, it works) * caller initiates a call * call is bridged to the SIP provider' side * we receive a T.38 Reinvite from the SIP provider' side * Freeswitch sends Trying back to the the SIP provider' side * Freeswitch logs: set(execute_on_answer=t38_gateway self) * Freeswitch sends a T.38 Reinvite to the caller, establishes T.38 on this side. * But the T.38 Reinvite from the SIP provider' side is not answerered * So the SIP provider' sidehangs up after 10 sec of timeout. What are we doing wrong here? Is it possible to act as a t.38 passthrough router and a T.38 gateway in the same dialplan? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Display Updates not sent from SIP to Verto Clients
by José Lopes 24 Feb '17

24 Feb '17

Hello guys, I have an issue with display updates from SIP to verto clients. The FreeSwitch receive SIP inbound call that goes to Verto client. This call is attended transfer on the caller side, that sends a SIP re-INVITE to FreeSwitch. This re-INVITE have a P-Asserted-Identity Header with updated display information. This information is not sent to verto client (verto.display event). Do I need to configure anything, so the display information at P-Asserted-Identity is sent to Verto client? Best regards, José Lopes

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GSM to GSM bridge with ringback question
by Anonim Stefan 24 Feb '17

24 Feb '17

Hi, I'm doing a GSM call bridge between two phones connected to a nanoBTS, with no dtmf. I am facing an issue only when setting ringback; here's my dialplan python code: bts_command="{ignore_early_media=true,bridge_early_media=false}sofia/gateway/gw/43112" session.answer() session.execute("set","ringback=local_stream://moh") -> plays the stream at 8000 Khz session.execute("bridge", bts_command) The problem is that when the B-peer answers, A-party ringback stops and I *can not* hear the B-party. If I'm doing this without the ringback line it works ok. See logs at [1] for this scenario. This is not happening with PCMA codec tests (which also have dtmf codec). I tried also libearal or disabled DTMF without success. I am using FreeSWITCH version: 1.6.12~64bit ( 64bit) Did something similar happen to someone before and has an idea how to fix this, while still keeping the ringback tone? [1] http://pastebin.com/XkcX7HXg Thank you, Stefan

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BYE not sent to caller
by David Villasmil 24 Feb '17

24 Feb '17

Hello all, I've got this situation where fs received a BYE from the bleg, on the cli it prints out "sending BYE to [channel]", but no BYE is actually sent. This is on TCP. This happened to anyone else? (Fs 1.4) Regards, David

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Re- FS-9821 mod_lua memory leak fix - will this be integrated with the production build at some stage?
by Andrew Keil 24 Feb '17

24 Feb '17

To FreeSWITCH users, Will the mod_lua memory leak fix (Jira FS-9821) be integrated with the production build branch at some stage ie. 1.6.x? Is there a current target production version? Regards, Andrew Keil Visytel Pty Ltd

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Max registers from single IP
by David Villasmil 23 Feb '17

23 Feb '17

Hello guys, Is there a way of setting a max number of registers from the same IP? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Class 5 Switch like voipswitch in Freeswitch
by Uday kumar 22 Feb '17

22 Feb '17

Hi All, Can you please help me to find out any opensource Class 5 switch developed with Freeswitch like voipswitch? -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

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Freeswitch server itself generating unwanted calls (Ghost calls)
by Shisheer Teli 22 Feb '17

22 Feb '17

Dear Team, from last few days my FreeSWITCH server generating some logs itself. I don't know how to stop this .. please help .. it's urgent. Please see the attached logs. -- Regards, Shisheer T *Save paper, save trees.... Please do not print this e-mail unless it is absolutely necessary.*

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AWS Instance Recommendations
by Colin Morelli 22 Feb '17

22 Feb '17

Hey list, I'll start by saying I have read the warnings and recommendations to *not* run FS on virtualized hardware. For right now, though, it's really the only option for me. I don't anticipate nor will I expect the behavior to be the same as what I would get on physical hosts, but would just like it to be as consistent (and good) as possible given the constraints. My question is - does anyone have particular kernel and instance type recommendations for AWS? I'm getting relatively frequent jitter and what seem like timing-related issues on calls, even in simple device <-> device calls within FS (no external gateway involved). I'm currently running FS in a Debian 8 docker container (with no memory or CPU limits) on a c4.large Ubuntu 16.04 host (kernel 4.4.0-45, default 250hz timer). The particular behavior I'm seeing *feels and sounds* like a few frames of silence, followed up by the audio being sped up slightly for a second or two (almost like it's trying to catch up), happening occasionally throughout calls. Note that this is with one call, and ~5% CPU usage on the host. Thus I'm pointing a finger most likely at kernel timer resolution. Running time_test 100 on FS returns an average of 200, which seems pretty bad. I've heard people have better success with the Amazon Linux AMIs for their packaged kernels. Is this true? Would recompiling the kernel with a 1000hz timer help a lot here? If you're having success on AWS, would be great to know what AMI you're based off of, what kernel version (and timer), and instance size. Thanks in advance. Best, Colin

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Conditional route-advance?
by Jean-Michel Hiver 22 Feb '17

22 Feb '17

Hello Freeswitch-users, In Freeswitch dialplan, you can route-hunt by stacking bridge commands like so: <extension name="internal"> <condition field="destination_number" expression="^1111"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/local_profile/1111@@" /> <action application="bridge" data="sofia/local_profile/1111(a)example2.company.com" /> </condition> </extension> My question is - is it possible to have more fine grain control about when to route-advance? For example, it might be OK to route advance when example1.company.com is not reachable or returns a 5XX response code or doesn't return a 183 Session progress or 200, but it may not be OK to route advance on a 4XX for instance... Any ideas on how to have finer control? I haven't found anything relevant in the docs, but if iI've missed something, pointers to the relevant doc would be highly appreciated. All the best JM

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