- FreeSWITCH-users - FreeSWITCH Mailing Lists

traffic and system monitoring
by robert mundkowsky 23 Jul '17

23 Jul '17

Is Homer the best free option for traffic and systemmonitoring GUI? Or isCACTI better option? Robert

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Path header handling
by Iskren Hadzhinedev 23 Jul '17

23 Jul '17

Hi, I have a kamailio proxy in front of FreeSWITCH. When a user registers, kamailio adds a path header like so: Path: <sip:KAMAILIO_PUBLIC_IP;lr;received=sip:UAC_PUBLIC_IP:5063> However, when FreeSWITCH sends the initial invite, it sends a Route header like this: Route: <sip:KAMAILIO_PUBLIC_IP>;lr;received=sip:UAC_PUBLIC_IP:5063 Is it possible to instruct FreeSWITCH to add the lr;received parameters inside the angle brackets as a part of the URI and how? e.g. how to achieve Route: <sip:KAMAILIO_PUBLIC_IP;lr;received=sip:UAC_PUBLIC_IP:5063> Thank you! Kind regards, Iskren Hadzhinedev

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Behaviour of blocking applications
by Oz Mortimer 22 Jul '17

22 Jul '17

Hi, I know I can work around this issue, but was wondering if the behaviour is expected… If I use an application that blocks, mod_http_cache for example and the web server is unavailable - the call will stay up until the connect-timeout is reached. Obviously I can set a lower connect-timeout, but this doesn’t really resolve the issue. So, say I set connect-timeout to 3 seconds and place a 1 second call via soft phone (caller hangup), the billable duration will be 3 seconds - should the billable duration not be 1 second to match the calling party? also, shouldn’t a hangup force any running applications to stop? Again, obviously I can avoid this by being a bit smarter with the usage of http_cache, etc - but was really wondering if a) thats expected behaviour and the callee should count the duration based on the ACK from the BYE (which I suspect non do), or b) maybe there is a duration variable that I’ve not yet seen that contains the call duration until we receive the BYE from caller?. I would have thought (and Im probably wrong), billable duration s/u/m should resemble the SIP durations, i.e. endtime = time BYE was received? The other byproduct of this symptom is that the channels will stack up until call-timeout is reached - if it was left to the default of 300 seconds - a dead web server could soon cause the switch to burn out. Again, easily resolvable by not using blocking applications, doing things in the background, etc but… Thoughts & postcards? Thanks Oz.

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Re: [Freeswitch-users] Multiple endpoints bridge issue
by 'Yemi Obembe 22 Jul '17

22 Jul '17

I totally understand that it should cancel all others. If Call A comes in and rings across user/xxxxx1,user/xxxxx2,user/xxxxx3 and xxxxx1 picks, I know ~2 and ~3 should be dropped. What I'd assume is that if Call B comes in (as Call A is engaged to xxxxx1), xxxxx2 and xxxxx3 should also be able to pick as they are available. As it is now, Call B will ring on them, but once they pick, as long as an existing call (A) exists, Call B drops. In other words, there can't be concurrent calls on the system. (What I am trying to create is a call center system where multiple agents can be engaged with different calls at the same time). Bridge is not a conferencing app. Bridge is used, normally within a dial plan to link an incoming a-leg call to an outgoing b-leg which is created to the endpoint’s specified in the bridge command. So the behaviour your experiencing is correct when specifying multiple endpoints, as soon as an endpoint/user answers the call the bridge command cancels all others. On 21 Jul 2017, at 13:04, 'Yemi Obembe <fickledreams(a)yahoo.com> wrote: I am using bridge via mod_httapi to call multiple endpoints via an xml response like this: <document type="text/freeswitch-httapi"><work> <execute application="bridge" data="user/xxxxx1,user/xxxxx2,user/xxxxx3" /> </work></document> What I notice is that this works for one or two connected endpoints. But once there are three or four endpoints, other incoming calls drop. I changed to enterprise originate (i.e, :_:) and noticed that once a call is engaged, other calls don't come in till that one ends. Same if I separate users with pipe (user/xxxxx1|user/xxxxx2|user/xxxxx3). Even when I bridge just a user at time, <execute application="bridge" data="user/xxxxx1" /> the other calls drop once one is engaged. What am I missing?

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Freeswitch call-ID configuration
by hemanth＠advaitamtech.com 21 Jul '17

21 Jul '17

Hello, I am using Freswitch as a media proxy and to handle SIP requests it is working as B2BUA, Here i am not able handle 407 response from server, Because Freeswitch is sending different Call-ID and From tag in the Invite carrying proxy-authorization header, Because of this behavior SIP server is sending 407 for Second Invite carrying Proxy-Authorization Header also, Can any one please help me to configure freeswitch, So that it sends Same Call-ID and from tag in Both the Invite requests Thanks & Regards Hemanth

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Understanding a dial string returned by group_call
by Mark Melling 21 Jul '17

21 Jul '17

When I execute a group_call command I get responses like this [^^:sip_invite_domain=example.com:presence_id=1010@example.com ]sofia/external/sip:1010@192.168.56.1:59527 ;ob;fs_nat=yes;fs_path=sip%3A1010%40192.168.56.1%3A59527%3Bob Are there any decent resources that would help me understand this better? As I understand it the [...] at the beginning means, apply variables to the channel, but what does the ^^ mean? And what does the bit after the address do/mean? i.e. in this specific case the bit ;ob;fs_nat=yes;fs_path=sip%3A1010%40192.168.56.1%3A59527%3Bob Thanks Mark

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Multiple endpoints bridge issue
by 'Yemi Obembe 21 Jul '17

21 Jul '17

I am using bridge via mod_httapi to call multiple endpoints via an xml response like this: <document type="text/freeswitch-httapi"><work><execute application="bridge" data="user/xxxxx1,user/xxxxx2,user/xxxxx3" /></work></document> What I notice is that this works for one or two connected endpoints. But once there are three or four endpoints, other incoming calls drop. I changed to enterprise originate (i.e, :_:) and noticed that once a call is engaged, other calls don't come in till that one ends. Same if I separate users with pipe (user/xxxxx1|user/xxxxx2|user/xxxxx3). Even when I bridge just a user at time, <execute application="bridge" data="user/xxxxx1" /> the other calls drop once one is engaged. What am I missing?

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Call forward
by Karlheinz Knapp 20 Jul '17

20 Jul '17

i have extension which is forwarded to a mobile number If I call this extension from a neighbor extension it works. But if I call from a outside number to the extension I don't get the voice on both sides. I believe it is a issue with reinvite. How I can fix this issue ? -- Diese Nachricht wurde von meinem Android-Gerät mit K-9 Mail gesendet.

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CDR - best practices
by robert mundkowsky 20 Jul '17

20 Jul '17

What are the best practice for writing Call DetailRecords (CDRs) to a database? We are using custom code to write CDR-like data to logfiles and then later loading logs into a database. We have very low usage right now, so Isuspect that we could just write directly to the database. But as we scale, is it a better idea to notwrite directly to the database? Note I am not sure if we need custom CDRs or if FS' CDRmodule might be leveraged. I have tolook into that. Robert

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group_call_function and record_session - Recording calls for a specific Extension within a call group
by Jonathan Hunter 20 Jul '17

20 Jul '17

Hi All, Hope everyone is well? I have a quick question around recording calls for extensions that are contained within a huntgroup. We use record_session to record calls when made outbound or inbound to specific extensions, however we do this using the bridge function. Is it possible to record calls for a specific extension contained within a call group? For example can dial-string be defined with record_session? Has any body done this before? Many thanks Jon

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