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send event to Grandstream phones when presence/dialog event has sent
by Tóth Róbert 26 Nov '10

26 Nov '10

Hi 1. I would like to send an event when presence or dialog events are sent. Because i have Grandstream phones, and i want to refresh their IdleScreens as it can be read the lower part of this page: http://www.voip-info.org/wiki/view/GXP-2000+XML+Idle+Screen Have i modify the code of sofia_presence.c or is there a simpliest way to send events bound to an other event? 2. And could you give an idea how can i insert records in the simpliest way to the database also when presence/dialog event sent? (Or is the simpliest to give code to sofia_presence.c?) Thank you Robert

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INVITE via another SIP proxy
by Ingmar Schraub 26 Nov '10

26 Nov '10

Hi, I ran into a problem today and I thought I should place the question here and hopefully someone has an idea what's going on. FYI, I am using GIT from November 15th 2010. Here is my problem: My SIP phones are all on the LAN. I can register them, make calls, etc. all work fine. I run FS with "-nonat". Now, when I add another SIP proxy in the chain, let my phones register with FS via that proxy, everything is still fine. No issues with registrations. The problem is, that INVITEs do not work. FS rejects them with "400 Bad Request". I turned on SIP tracing on FS to see how the SIP messages look like and I noticed that FS is for some reason stripping off (apparently..) the SDP part from the INVITE. Why do I believe that? Well, I used tcpdump on the FS box to capture the network traffic and see what arrives actually. The INVITE received by the network stack shows the full INVITE message including the SDP part. On the other hand, when I make a call / INVITE from a phone which is directly registered at FS, the INVITE message shown in the sip trace includes the SDP. Thus, I believe that FS has an issue parsing that particular INVITE message properly. Note: the same proxy, the same set-up works all sorts of other PBXs and Soft-Switches (Asterisk, Avaya, 3com, Nortel, Cisco, Sonus, ...). It's all tested. Of course I can provide the FS SIP trace log and the corresponding tcpdump. If any developer could have a quick look, I'd really appreciate this. Thanks and best regards, Ingmar

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Load Balancing using OpenSIPS and B2BUA
by Abid Saleem 26 Nov '10

26 Nov '10

Hi All, What is the best load balancing method to use with FS. I think using OpenSIPS method seems to be better. Could anyone tell me who will work as a B2BUA in this setup OpenSIPS load balancer server or individual FS boxes? In case we use OpenSIPS as the B2BUA which is not a full B2BUA and does not handle media, so how this scenario should be best configured to provide optimum performance. Suggessions, recommendations and experiences are required. Thanks Regards----------Abid SaleemTechnical Manager NGNTerminus Technologies

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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-----

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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.

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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?

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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

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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.

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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:

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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

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