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evaluate
by mbo 23 Jan '14

23 Jan '14

Hello, in a bridge I can evaluate a sofia contact like <action application="bridge" data="${sofia_contact(internal/100(a)sys.local)}^222" /> that works fine. If I try to use the same approach in a originate like: originate ${sofia_contact(user/1000)} &park I can see: 2014-01-23 20:14:18.216587 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables 2014-01-23 20:14:18.216587 [ERR] switch_core_session.c:496 Could not locate channel type ${sofia_contact(user 2014-01-23 20:14:18.216587 [NOTICE] switch_ivr_originate.c:2651 Cannot create outgoing channel of type [${sofia_contact(user] cause: [CHAN_NOT_IMPLEMENTED] 2014-01-23 20:14:18.216587 [DEBUG] switch_ivr_originate.c:3622 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] Is there a way to resolve/evaluate sofia_contact also in an originate? Thanks Markus

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Change default extensions
by peetee23＠hushmail.com 23 Jan '14

23 Jan '14

Dear All, could you please explain me what I have to change in dialplan to use SIP extensions 1-19 instead of default 1000-1019? Above of that what is essential in dialplan to have working internal SIP calls and maybe SMS/SIMPLE messages only? Thanks a lot for any feedback! Kind regards, PT

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freeswitch in sandbox
by Joli Martinez 23 Jan '14

23 Jan '14

Hello, I have a python script located in /usr/local/freeswitch/scripts and a lua script calls the python script. Scripts works when calling lua <scriptname>, but when invoking it from freeswitch I get "file not found" on the python script. permissions are set to 755 and owner is freeswitch:daemon. both the Lua and the python script are in the same scripts directory. The python script makes a mysql call and parses some data and return it back to Lua. I remember having this problem before and I believe that freeswitch was in a sandbox enviroment, but dont remember what I changed. Thanks,

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Caller ID unknown with Sangoma A200
by Peter Hartmann 23 Jan '14

23 Jan '14

Hi, Is there anything special I need to do to enable caller id on a Sanagoma A200 analog card? Everything is working with this card except for incoming caller id. I have <param name="enable-callerid" value="true"/> in the fxo section of my conf/autoload/freetdm.conf.xml. By the way, I have also tried underscore....I've seen it both ways on Sangoma's site and the wiki. I put a snippet of the log here: http://pastebin.com/nAsr0wQj There's plenty mention of GET_CALLERID but when it comes time to refer to the caller it always comes up 'unknown' like so. [INFO] mod_dialplan_xml.c:558 Processing unknown <unknown>->212XXXXXXX in context public By the way, this is using Version 1.5.8b git a26fdca, and this is in the NYC area. I'm assuming we get the fsk between rings one and two. I though it's curious that I haven't had to specify what type of caller id I'm using in wanpipe via wancfg_fs, wancfg or in FS. I did see references to bell202 and other types in the wanpipe api sources. Does FS tell wanpipe to use Bellcore by default? Thanks much in advance, Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com

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rtp-timer-name none packet loss handling
by Jozsef Vass 23 Jan '14

23 Jan '14

We are using freeswitch to transcode between opus and g711. We have rtp-timer-name set to none in sip_profiles. If we experience packet loss on the inbound leg (opus), freeswitch will take no action. When the next opus packet arrives, freeswitch sends a pcmu rtp packet with continuous sequence number and timestamp to the outbound leg. That is, there is no way to tell from the outbound leg that there was a packet loss on the inbound leg. I would prefer the following behavior: 1. When there is a packet loss on the inbound leg, increase the sequence number and timestamp on the outbound leg. 2. When there is a packet loss on the inbound leg, interpolate the missing packet using opus plc. This only works for a few consecutive lost packets and not for burst loss. Is there any config parameters to force either of the above behavior? Jozsef

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disable user features in conference
by Joli Martinez 23 Jan '14

23 Jan '14

Hello, When a user logs into the conference he is able to press * to mute, # to kick himself out, and other options. I would like to disable them all and let me manage the state of the user. Where would I disable those options? Thanks,

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call rejected due to min_idle_cpu overload counter
by Alain BIEUZENT 23 Jan '14

23 Jan '14

Hi All, is there a way to know the number of calls that have been rejected by exceeding the value of min_idle_cpu ? Regards Alain.

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trans-coding is not working
by Shahzad Bhatti 23 Jan '14

23 Jan '14

Hi Everybody, i want to call legB in a scnerio that i have "inbound-codec-prefs" value="GSM,PCMA,PCMU,G729" "outbound-codec-prefs" value="PCMA,PCMU,G729" so i change sip profile as <param name="inbound-zrtp-passthru" value="false"/> <param name="inbound-late-negotiation" value="true"/> then after *reload mod_sofia* i originate call to bLeg where on inbound i only offer GSM Codec and i think that GSM codec should trans code using PCMA but i got response [415] here is my cli log http://pastebin.freeswitch.org/21881 tell me how i can trans code and correct me if i am wrong; Regards Shahzad Bhatti

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no video when bridge call to PBX ACD
by Jimmy Chang 23 Jan '14

23 Jan '14

Hi folks, I've made a video call from a SIP softphone to a PBX ACD queue that was bridged by FS. I set the PBX(AASTRA) as a gateway in FS. Here is the scenario. Softphone --> FS --> PBX --> IVR |---> Agent SIP phone The call can be established with no video. If I make the call bypass FS like this: Softphone --> PBX --> IVR |---> Agent SIP phone The video call can be make successfully. All these device are all behind NAT. I set the parameters renegotiate-codec-on-reinvite=true, inbound-late-negotiation=true and still no video. I can't figure out why the video missed. Here is the FS siptrace. http://pastebin.freeswitch.org/21879 Thanks in advance. Jimmy --- 由於 avast! 防毒軟體已開啟防護，所以此封電子郵件將不受病毒及惡意軟體的侵害。 http://www.avast.com

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detecting if user available before bridge
by ik 23 Jan '14

23 Jan '14

Hello, I need to find if a user is connected to FS currently. That is, not only registered (I found how to detect that<https://wiki.freeswitch.org/wiki/Function_sofia_contact>), but for example if I can communicate with it's soft-phone. I need to know if the soft-phone crashed for example, and if not, then not to send the bridge command at all, but actually been able to do something back in my application because of it. Is there a way to send for example an "option" header of some sort to a specific user prior to bridge command, and know it's result ? Thanks, Ido

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