November 2012 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

About the display problem during calling
by sparklezou 05 Dec '12

05 Dec '12

Hi All, I facing a strange problem. Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine. But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number. I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem. I checked the sip info. The "Outbound Call" is in the 200 SDP message. the called is set to ""Remote-Part-ID: "Outbound Call" <sip: 1234(a)a.b.c.d>; ....... " Not the correct called name. Is there anything wrong? Please help me. Thanks! 2012-11-30 sparklezou

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Conference CDRs via http
by Mariusz Czulada 04 Dec '12

04 Dec '12

Hi all, I was thinking about sending CDRs via HTTP in a same or similar way mod_xml_cdr does. I consider implementing this (unless someone else is working on it) but first wanted to discuss with you the best approach. A. All in mod_conference This would require to copy many fragments of code from mod_xml_cdr into mod_conference. Also same configuration parameters used in xml_cdr must be processed and used for sending data. The advantage for this solution is that everything related to this mechanism is included in this module. Drawback: if something in mod_xml_cdr requires fixing or extending, probably the same changes should by applied in related parts of mod_conference. B. mod_conference builds, mod_xml_cdr sends data For this solution I'll give more details. 1. New event type should be added (like SWITCH_EVENT_CDR) 2. When a module (in this case: mod_conference) wants to store CDR via HTTP it must fire an event of that type and: - "Event-Subclass" set (like "conference") - Only common headers are needed, plus... - "Content-Type" and "Content-Length" must be set - CDR data must be build as XML in a module and added as an event content. 3. mod_xml_cdr will listen to this event type. 4. For each event subclass which mod_xml_cdr must must react, configuration file will contain a set of params same as for generic channel CDRs. 5. If subclass matches configuration, mod_xml_cdr reads data from event content and sends them according to configuration. Changes in mod_conference: - one new parameter for each profile (like "cdr-via-event=yes|no") - if "yes" then xml must be build even if "cdr-log-dir" is unset - if "yes", then an event must be fired as described above Changes in mod_xml_cdr: - extra parameters from configuration to be parsed (like '<settings subclass="conference">....</settings>') - bind to SWITCH_EVENT_CDR - if 'Event-Subclass' matches configuration, a content of the event will be sent via HTTP (probably most of 'my_on_report' routine bellow 'try to post it to the web server' comment will be reused) A [small] drawback is that it makes an indirect module dependency, but we already have such situations (like mod_shout needed to record a conference in mp3). Advantages are: - one can create an external tool for handling this type of event (to store it directly in db or send it with other protocols) - this mechanism can be easily reused in other modules if needed; maybe in mod_callcenter, maybe in other components. No further changes in mod_xml_cdr should be needed. What is unknown to me is a maximum size of event content. Conference CDR XMLs can be quite big - will it be a problem to send it this way? I think the second solution is better and more universal but I'd like to hear your opinions about this case. Regards, Mariusz

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About the "transfer" function
by sparklezou 29 Nov '12

29 Nov '12

Hi All, When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321". And on the caller phone will disply "Outbound Call", also show the number "87654321". I checked the sip message, the info is got from the 180 or 183 message. Sometime, don't want to show the real external number "87654321" to the enduser. How to keep the caller phone displaying "1234". Thanks in advance! 2012-11-28 sparklezou

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rpm problem
by Michel Daggelinckx 29 Nov '12

29 Nov '12

Just updated my test system and got Package freeswitch-application-sms-1.2.5.1-1.el6.i386.rpm is not signed Michel

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About module "Mod_pandsp"
by sparklezou 26 Nov '12

26 Nov '12

Hi Mod_spandsp developers, Is it possible to develop Mod_spandsp like "Mod_voicemail"? I means, set all of the related parameters in the user configure file. such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail sender configuration, such as SMTP. Something seemd the same, send "voice message" via eamil. Send "fax file" via email. Thanks! 2012-11-26 sparklezou

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rtcp-mux support
by openser 26 Nov '12

26 Nov '12

Hi all, Does freeswitch support rtcp-mux feature ? if it support , freeswitch should send rtcp packet by rtp port but not rtp +1 for given audio/video session, my client support rtcp-mux , but freeswitch send rtcp to port rtp +1 .

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FreeSWITCH Weekly News and Notes
by Michael Collins 26 Nov '12

26 Nov '12

Hello all! We are all back to a full week after many of us enjoyed some well-deserved time off last week. However, even though there was a holiday here in the US, the intrepid FreeSWITCH development team was working hard on your behalf. As Ken Rice previously mentioned, Anthony spotted a potential issue in the recently released 1.2.5 version. Therefore, this past Saturday they made 1.2.5.1 available<http://files.freeswitch.org/freeswitch-1.2.5.1.tar.bz2>for us. Many thanks to those who work so hard to make sure that FreeSWITCH is running smoothly for us all. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>we spent some time getting everyone up to speed on how to edit the FreeSWITCH wiki <http://wiki.freeswitch.org>, specifically focusing on channel variables <http://wiki.freeswitch.org/wiki/Channel_Variables> pages. Updating documentation is one of the least glamorous aspects of maintaining an open source project. Many thanks to those who've stepped up over the past weeks and months to help us out. With the end of the year upon us we are slowing down a bit in our speaking schedule for the weekly community conference call. We have a few things in the works but nothing yet scheduled. On this week's call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>we will be doing a community scrum. Be sure to bring your questions and topics for discussion. If you have a tip or trick that you'd like to share with the group that would be most welcomed. If time permits we will crowdsource a few selected questions from the mailing list. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 77, Issue 6
by David Rot 26 Nov '12

26 Nov '12

Dears 1. In some occasion I get the caller-id with an extension, and in other occasions I get the caller-id without an extension. How do I make sure that I always get the caller-id with an extension? 2. Soon after I perform a recording, I hear a tick sound (like hanging up) and then the call continues without any problems. This situation is very confusing for our customers as they believe that the call was disconnected (because of the tick sound). How can I avoid the tick sound that causes the confusion? Thanks

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Failed DTMF sanity check.
by Debasish Chandra 26 Nov '12

26 Nov '12

Hi, I am getting "Failed DTMF sanity check" randomly (not on every call, 2 - 3 calls out of 10). Application is running using Plivo REST API, while FS receiving call from Sangoma NSG TDM gateway. I have read on FS wiki that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end' packet. However call tcpdump shows that DTMF digits has been sent to FS from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores the digit. Please let me know if I need to do any extra settings like editing dtmf duration or type. Best Regards, Debasish Chandra

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Redis Module Implementation
by Rodney 26 Nov '12

26 Nov '12

I'm currently looking at the redis module as a way of implementing limits that are shared across multiple servers and noticed that for every limit I create, two keys are set in redis. One key is what I'm expecting, with the name derived from the values I've supplied, however the other key is the same thing but with the hostname prefixed. Looking at the code, this could be either the hostname or the switchname that is being prefixed (depending on if a switchname is set). My question is basically why is it doing this ? I would have assumed that if a dialplan author wanted the key in question to have the hostname or switch name, couldn't they just include it in the call to limit in the dial plan ? It also looks to me like there could be a bug where the hostname prefixed key only gets incremented if there is a max set but will always get decremented. I'm just trying to work out if this is intended behaviour and I have just misunderstood the purpose of the module, or if it is worth trying to put a patch together. Thanks, Rodney

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