January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Script for Presence using PHP
by Edmar Cruz 07 Jan '10

07 Jan '10

Dear Sir, Is there a script to trigger whether a user is online or offline using a script or a PHP code? Presence for freeswitch is for command line only? Can you give me an example? Thanks, Edmar -- View this message in context: http://old.nabble.com/Script-for-Presence-using-PHP-tp27050100p27050100.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Codec Negotiation: Codec telephone-event:101:8000:20
by Mark Campbell-Smith 07 Jan '10

07 Jan '10

Hi! If I try to call out on one of my voip providers I get INCOMPATIBLE_DESTINATION. Something is going wrong with codec negotiation: 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3262 Set 2833 dtmf payload to 101 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [NOTICE] sofia.c:3937 Hangup sofia/external/020555500 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] According to their website, the codec they us is G711 a-law or G729. What codecs have name CN and telephone-event? How do I get these to match? I assume the format of the debug output is: Audio Codec compare [Codec Name:Media Format:Media Name:Rate:RTP Size?] Thanks

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Is there support for custom fields for all events
by Jon Bruel 06 Jan '10

06 Jan '10

I would like to add a custom field to all events sent from the FS. The value of the field should be the value of a global variable, the name should preferably be the name of this variable. Is there any way to set this up? /Jon

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re lease an outbound call when caller sends digits like ##
by Oscav 06 Jan '10

06 Jan '10

Hi, How can we cancel an outbound call if the caller digits some DTMF like ## ?? There is the bind_meta_app but it only handles 1 digit. Thanks. -- View this message in context: http://old.nabble.com/release-an-outbound-call-when-caller-sends-digits-lik… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Dahdi Saga continues
by Joseph L. Casale 06 Jan '10

06 Jan '10

So when the dahdi config is left empty, just defaultzone=us and loadzone=us, it works except obviously there is no echo canceller. Once I add an echo canceller and set fxsks=1 then callerid fails, it just says OpenZap on the handsets and the audio doesn't start working after some time after the call is answered. Any ideas? Thanks! jlc

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Re: [Freeswitch-users] Detecting status of User Agent
by Jon Bruel 06 Jan '10

06 Jan '10

Hi Durk I tried the situation mentioned by unplugging the UA. FS then reports back (via the channel variable originate_disposition): NORMAL_TEMPORARY_FAILURE, which can then be used to take action. Before bridging to the phone, I have set <action application="set" data="continue_on_fail=true"/>. Hope it assists you. Anyhow, your cat must be very intelligent since it knows it's a CAT cable. /Jon Jon Brüel Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ________________________________ From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Durk.de Beer Sent: 6. januar 2010 10:04 To: Freeswitch-Users Subject: [Freeswitch-users] Detecting status of User Agent Ok I want to do the following thing. If an user agent (UA), some SIP-client for instance X-lite, is of line the call is to be forwarded to an phone number provided by the user how was called. I succeed doing this if there is no registration for the number dialled. FS is then reporting an USER_NOT_REGISTERED so I am able to alter the dial plan accordingly and redirect the call. So far so good. Now my problem arises when there is an registration on FS but the UA is not online for what ever reason (my cat seems to like to chew on CAT5 cable). If this occurs FS is trying to bridge but it is unable to because there is no response from the UA called. It ends by a RECOVERY_ON_TIMER_EXPIRE but the time it takes for this to happen is far to long. I've tried to set this variable but failed to do so. Other channel variables seem to be only effective if there's an successful bridge. So how do I set the RECOVERY_ON_TIMER_EXPIRE? Durk

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Detecting status of User Agent
by Durk.de Beer 06 Jan '10

06 Jan '10

Ok I want to do the following thing. If an user agent (UA), some SIP-client for instance X-lite, is of line the call is to be forwarded to an phone number provided by the user how was called. I succeed doing this if there is no registration for the number dialled. FS is then reporting an USER_NOT_REGISTERED so I am able to alter the dial plan accordingly and redirect the call. So far so good. Now my problem arises when there is an registration on FS but the UA is not online for what ever reason (my cat seems to like to chew on CAT5 cable). If this occurs FS is trying to bridge but it is unable to because there is no response from the UA called. It ends by a RECOVERY_ON_TIMER_EXPIRE but the time it takes for this to happen is far to long. I've tried to set this variable but failed to do so. Other channel variables seem to be only effective if there's an successful bridge. So how do I set the RECOVERY_ON_TIMER_EXPIRE? Durk

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Migrating from asterisk to FS
by Nik Middleton 06 Jan '10

06 Jan '10

HI Guys While I've been using FS for around 18 months now, and love it to bits, it's been a specific solution. I'm now looking to move my customer base across, and have on the base of it some basic and perhaps dumb questions. I currently have around 150 Sip phones attached to my systems These are all geographically spread, so re-configuring them is out of the question. They all register on port 5060. Given that FS uses port 5080 for external clients, do I simply need to do a Port translate on my firewall or is there a simpler solution? Further, how does FS handle a call FWD? In other words, if a SIP phone has a divert on busy set will it account for the redirect? Currently in Asterisk I use the 'I' option to disable this as I can't account for the call. Regards,

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Zap channel not released when voicemail starts
by François Legal 06 Jan '10

06 Jan '10

Hello, I have an issue with voicemail and openzap channels. When an incoming call on an openzap channel is bridged to voicemail, if that channel is hung up before the beginning of voicemail recording, that channel is kept open open until 3 or 4 seconds after the voicemail started to record the message. What should I do to make freeswitch/voicemail release the channel immediately when the caller hang up ? Thanks in advance François

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unable to call out troug siemens hie9200 in pur SIP
by Ron McLeod 05 Jan '10

05 Jan '10

Posted on behalf of Tayeb Meftah... hi dear friends, we have a siemens hie9200 softswitch we want to interconnect freeswitch with it to use it for service, like media, voicemail, audio conferencing and ... call trace is atached belo if we calls from tdm, call is passed but without rtp i routed a number like 021000000 is a tdm number to 3001 the default 8khz fs conference so inbound call is passed but without audio for outbound calls, i wanted to call a tdm number, like 021298235 the call tack some long time but return a 500 internal server error from the hie9200 softswitch: sip.Reason == "Q.850 ;cause=47 ;text=\"Resource unavailable, unspecified\"" from fs i get normal temporary failur so please see the trace and return for me any reply ;) thanks for your helps and times. Traces at: http://siplabs.net/tracebin/fs-siemens-500.pcap http://siplabs.net/tracebin/fs-siemens-rtp.pcap

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