November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

dialog info missing in presence message when using originate api command
by Helmut Kuper 25 Nov '10

25 Nov '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have an application and internal phones A, B and C. Phone B is subscribing for presence information of phone A. If I switch phone C to phone A via xml dialplan and bridge app, phone B is informed by FS via a SIP NOTIFY containig something like this: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="8" state="full" entity="sip:4912@85.16.246.6"> <dialog id="673cea17-58ac-4c93-a90d-c1e9eb34a8cd" direction="recipient"> <state>early</state> <local> <identity display="4912">sip:4912@85.16.246.6</identity> <target uri="sip:4912@85.16.246.6"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="2850">sip:2850@85.16.246.6</identity> <target uri="sip:**4912@85.16.246.6"/> </remote> </dialog> </dialog-info> So phone B displays the ringing call of phone A in its display If I call phone A with api command originate with this dial-string: execute_on_ring=info,origination_uuid=111111111111111,originate_timeout=15,origination_caller_id_name=dddd,origination_caller_id_number=1234,ignore_early_media=true then phone A is ringing, but phone B display nothing. The SIP NOTIFY to phone B ist contains this: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:4912@85.16.246.6"> <dialog id="88abe007-278c-4334-a309-0b9b8c2f4a72" direction="recipient"> <state>early</state> </dialog> </dialog-info> In sofia_presence.c it seems that neither from_id nor from_user is set. Both are extracted from the presence event The presence event is this: 2010-11-25 21:28:21.096675 [INFO] sofia_presence.c:1399 CHANNEL_DATA: Event-Name: [PRESENCE_IN] Core-UUID: [1a9afaec-418b-4586-9235-502c51f758f3] FreeSWITCH-Hostname: [ippbx-prod] FreeSWITCH-IPv4: [85.16.246.6] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2010-11-25 21:28:20] Event-Date-GMT: [Thu, 25 Nov 2010 20:28:20 GMT] Event-Date-Timestamp: [1290716900357839] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_perform_presence] Event-Calling-Line-Number: [584] Channel-State: [CS_ROUTING] Channel-Call-State: [DOWN] Channel-State-Number: [2] Channel-Name: [sofia/internal/sip:4912@85.16.245.243:5060] Unique-ID: [25f99412-5291-472f-9c1a-2c6944d5b294] Call-Direction: [outbound] Presence-Call-Direction: [outbound] Channel-Presence-ID: [4912(a)85.16.246.6] Channel-Call-UUID: [0cee2aa5-e1b1-4d9b-9ec7-89071351259f] Answer-State: [ringing] Caller-Direction: [outbound] Caller-Caller-ID-Name: [ mip (2850)] Caller-Caller-ID-Number: [2850] Caller-Destination-Number: [4912] Caller-Unique-ID: [25f99412-5291-472f-9c1a-2c6944d5b294] Caller-Source: [src/switch_ivr_originate.c] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/sip:4912@85.16.245.243:5060] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1290716900357839] Caller-Channel-Created-Time: [1290716900357839] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] proto: [src/switch_channel.c] login: [src/switch_channel.c] from: [4912(a)85.16.246.6] rpid: [unknown] status: [CS_ROUTING] event_type: [presence] alt_event_type: [dialog] presence-call-direction: [outbound] event_count: [0] Presence-Calling-File: [src/switch_channel.c] Presence-Calling-Function: [switch_channel_perform_set_running_state] Presence-Calling-Line: [1607] Any trick to get it working except hacking the code? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkzuyMoACgkQ4tZeNddg3dwOXwCfUAr1qi4eWfUlNMgGwIB5ctwk 9UkAoLp6gNl8ou5NsateSJuMfHVrJ6G8 =XApQ -----END PGP SIGNATURE-----

1 0

Regex "local Extension" is not mached
by Isti 25 Nov '10

25 Nov '10

Hi, does anybody know why a local extension is not matched by ="(^\d{3,6}$)">? All the extension are set to "default". It should generally work. Thank you for the help. Regards, I.

3 2

Lua and dbh:query returning multiple rows
by elijah 25 Nov '10

25 Nov '10

Could you demonstrate how to iterate through multiple returned rows in Lua from the function Dbh:query?

2 3

more info about mod_stress
by mashudi72 - 25 Nov '10

25 Nov '10

Dear All, Does any one have any information about mod_stress? regards, Mashudi

2 1

pocketsphinx grammar error
by ovvenkat 25 Nov '10

25 Nov '10

Hi to all, I am doing little research on pocketsphinx ASR. If I include Any name (example: venki) in the grammar file, I am getting "*Error loading Grammar*" . Its working fine with, "Yes" or "No" Anyone plz shed me some light on this? -- Regards Venkatesan OV.

1 0

Re: [Freeswitch-users] Regex "local Extension" is not mached
by Nyamul Hassan 25 Nov '10

25 Nov '10

If you look at the debug log level 7 from the fs_cli, then it tells you at what sequence the dialplan was processed, and what finally matched. Regards HASSAN On 25 Nov 2010 12:33, "Isti" <pistike040(a)googlemail.com> wrote:

1 0

Looking for more info about mod_rtmp
by Jarle Aase 25 Nov '10

25 Nov '10

I'm looking for a way to bridge rtmp to sip / audio-conferences, and came across a reference to "mod_rtmp". Does anyone have any information about that project? Jarle

4 3

propogating channel variables
by elijah 25 Nov '10

25 Nov '10

There is a channel variable I am adding to calls as they arrive that I would like to see within raised events. Specifically, I am adding my own database index to the A-leg, passing the call to mod_callcenter, then capturing the CHANNEL_ANSWER event where I am looking for this same index value. Unfortunately, I am not able to persist this custom channel variable for the duration of the call, despite using ‘export’. Could you suggest how I may accomplish a channel variable’s persistence in this situation?

4 10

Call-State in LUA
by Tóth Róbert 24 Nov '10

24 Nov '10

Hi, Is it possible to get a channel's Call-State in LUA? Is there a function for this similar to session:getState()? (I would like to know if a channel is in HELD state.) Thanks

2 1

Possibility to remove/disable Allow-Events header?
by Peter Olsson 24 Nov '10

24 Nov '10

I have a problem when communicating to an Avaya Communication Manager PBX, it seems to fail if the Allow-Events header is present. I've reported this as a bug to Avaya, but I just wanted to make sure if it's in any way possible to stop FS from sending this? I understand if it's not, since this is really an Avaya problem - but I just wanted to make sure, before I start poking around in the code.. :) /Peter

1 0