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Re: [Freeswitch-dev] mod_skypiax inching forward
by seven du 26 Jan '09

26 Jan '09

hi Giovanni, I just made some improve on skypiax, so you can call skypiax/ANY/ another_skypename, it will automatically chose an available channel. not good, but it works. put the following code directly before the following line: for (i = 0; i < SKYPIAX_MAX_INTERFACES; i++) { And change the above line to for (i = 0; !found && i < SKYPIAX_MAX_INTERFACES; i++) { It should be like: if (strncmp("ANY", interface_name, strlen(interface_name)) == 0) { //find an available one, allowing call like originate skypiax/ ANY/another_skypename DEBUGA_SKYPE("Finding one available skype interface\n", SKYPIAX_P_LOG); for (i = 0; !found && i < SKYPIAX_MAX_INTERFACES; i++) { if (strlen(globals.SKYPIAX_INTERFACES[i].name)) { int skype_state = 0; tech_pvt = &globals.SKYPIAX_INTERFACES[i]; skype_state = tech_pvt->interface_state; DEBUGA_SKYPE("skype interface: %d, name: %s, state: %d\n", SKYPIAX_P_LOG, i, globals.SKYPIAX_INTERFACES[i].name, skype_state); if (SKYPIAX_STATE_DOWN == skype_state || 0 == skype_state) { found=1; break; } } } } for (i = 0; !found && i < SKYPIAX_MAX_INTERFACES; i++) { Regards, Seven

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Trouble registering SIP client
by Roland Zagler 25 Jan '09

25 Jan '09

Hi everybody, I am writing a Java SIP ua (based on mjsip 1.6) that is supposed to register to freeswitch. Unfortunately I am not able to register and I cannot find the problem, although a softphone (X-Lite) CAN register by using the same creds. The credentials are: SIP username: roland SIP password: roland SIP realm: 10.0.4.60 There is no acl activated. The freeswitch versions I have tested are 1.0.2 and the trunk of today (11480). I hope someone on the list could take a look on the trace I attached below and give me a hint what goes wrong. Thank you very much in advance. Roland tport_wakeup_pri(0x81042c8): events IN tport_recv_event(0x81042c8) tport_recv_iovec(0x81042c8) msg 0xb3415ed8 from (udp/10.0.4.60:5060) has 373 bytes, veclen = 1 recv 373 bytes from udp/[10.0.2.51]:32816 at 17:01:30.461289: ------------------------------------------------------------------------ REGISTER sip:10.0.4.60:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562 Max-Forwards: 70 To: "roland" <sip:roland@10.0.4.60:5060> From: "roland" <sip:roland@10.0.4.60:5060>;tag=z9hG4bK84430669 Call-ID: 218108472980(a)10.0.2.51 CSeq: 1 REGISTER Contact: <sip:roland@10.0.2.51:32816> Expires: 3600 User-Agent: ZaKoSIP v1.2 Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x81042c8): msg 0xb3415ed8 (373 bytes) from udp/10.0.2.51:5060/sip next=(nil) nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 1) nta: canonizing sip:10.0.4.60:5060 with contact nta: REGISTER (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80fe270, 0x80f9448, 0xb3402b30) called soa_set_params(static::0xb341ba78, ...) called nua: nua_application_event: entering nua: nua_respond: entering nua(0xb3402b30): sent signal r_respond 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:883 sofia_reg_handle_register() Requesting Registration from: [roland(a)10.0.4.60] nua: nua_handle_destroy: entering nua(0xb3402b30): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering soa_set_params(static::0xb341ba78, ...) called tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816 tport_resolve addrinfo = 10.0.2.51:32816 tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816 tport_vsend returned 620 send 620 bytes to udp/[10.0.2.51]:32816 at 17:01:30.586602: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562 From: "roland" <sip:roland@10.0.4.60:5060>;tag=z9hG4bK84430669 To: "roland" <sip:roland@10.0.4.60:5060>;tag=aZa7matyy5Uca Call-ID: 218108472980(a)10.0.2.51 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="10.0.4.60", nonce="a5608fcc-ea38-11dd-8437-6d678b851040", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (1) nta: timer set to 32000 ms nta_leg_destroy((nil)) soa_destroy(static::0xb341ba78) called tport_wakeup_pri(0x81042c8): events IN tport_recv_event(0x81042c8) tport_recv_iovec(0x81042c8) msg 0xb3444050 from (udp/10.0.4.60:5060) has 575 bytes, veclen = 1 recv 575 bytes from udp/[10.0.2.51]:32816 at 17:01:30.614376: ------------------------------------------------------------------------ REGISTER sip:10.0.4.60:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562 Max-Forwards: 70 To: "roland" <sip:roland@10.0.4.60:5060> From: "roland" <sip:roland@10.0.4.60:5060>;tag=z9hG4bK84430669 Call-ID: 218108472980(a)10.0.2.51 CSeq: 2 REGISTER Contact: <sip:roland@10.0.2.51:32816> Expires: 3600 User-Agent: ZaKoSIP v1.2 Authorization: Digest username="roland", realm="10.0.4.60", nonce="a5608fcc-ea38-11dd-8437-6d678b851040", uri="sip:10.0.4.60:5060", algorithm=MD5, qop=auth, response="2f26d787c68ac2a342a4b1d7bb49a1a0" Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x81042c8): msg 0xb3444050 (575 bytes) from udp/10.0.2.51:5060/sip next=(nil) nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 2) nta: canonizing sip:10.0.4.60:5060 with contact nta: REGISTER (2) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80fe270, 0x80f9448, 0xb3445690) called soa_set_params(static::0xb341ba78, ...) called nua: nua_application_event: entering 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:869 sofia_reg_handle_register() Send challenge for [roland(a)10.0.4.60] nua: nua_respond: entering nua(0xb3445690): sent signal r_respond nua: nua_handle_destroy: entering nua(0xb3445690): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering soa_set_params(static::0xb341ba78, ...) called tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816 tport_resolve addrinfo = 10.0.2.51:32816 tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816 tport_vsend returned 500 send 500 bytes to udp/[10.0.2.51]:32816 at 17:01:30.617125: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562 From: "roland" <sip:roland@10.0.4.60:5060>;tag=z9hG4bK84430669 To: "roland" <sip:roland@10.0.4.60:5060>;tag=B83Zp5a2UejZN Call-ID: 218108472980(a)10.0.2.51 CSeq: 2 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 403 Forbidden for REGISTER (2) nta_leg_destroy((nil)) soa_destroy(static::0xb341ba78) called

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switch_frame structure question
by ludovic 22 Jan '09

22 Jan '09

Hi, what is the difference in the structure switch_frame between the following members : packet data payload The routine channel_write_frame of my endpoind is called, but it seems that I do not use the correct data as my payload_type of the data is not 8 (PCMA) which I should have. I call from a SIP phone, registered in freeswitch to an analog phone. I have to pass the RTP packet to my phone device. Thanks, Ludovic

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New wiki page needs your attention!
by Michael Collins 21 Jan '09

21 Jan '09

Hello FreeSWITCHers! I'm putting the finishing touches on a wiki page that we hope will make it easier for users to request help and for the dev team and power users to digest and process those requests. What I need first and foremost is for everyone to please read this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please give me feedback. Put yourself in the shoes of a relative newbie. Is the information easy to follow? On the flip side, if you wanted to help someone, ask yourself, if they follow the steps on this page will that suffice? Are there places that need improvement? Can you think of anything else that can be added? NOTE: I'm still working on the TDM/OpenZAP section as well as the sections on scripting, event socket, elements of a jira ticket, etc. If you have suggestions for content on those pages please email me off list or hop and and fill in some of the blanks. The core development team really appreciates all of your help. Now that FS is growing like mad we are at the point where it is imperative that we have reliable documentation for new ones so that the developers and other experts can focus on advancing the project even further. Let's all lend a hand by improving the documentation. Thanks again! -MC (mercutioviz)

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DTMF events
by Rob Charlton 21 Jan '09

21 Jan '09

Hi, I'm using mod_event_socket to listen for DTMF events. I have Nokia handsets registered as SIP clients over Wifi, as well as a SIP trunk providing incoming PSTN calls to a range of DDIs and outgoing PSTN calls. If I make an incoming (PSTN or SIP) call and answer it, I always see DTMF events via mod_event_socket. If I make an outgoing call direct to a handset using SIP then I see DTMF events - e.g. originate user/1000 &park() If I make an outgoing call via PSTN then I don't see DTMF events e.g. originate sofia/gateway/mygateway/myphonenumber &park() or &javascript(myscript.js); In the latter case, I am still able to pick up DTMF digits if I use javascript session.collectInput() - so it appears as if the DTMF tones are being recognised by Freeswitch, but no events sent. What am I doing wrong? Cheers Rob -- Rob Charlton Savage Minds Ltd

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Reg: Freeswitch execution error [urgent]
by Sanju 21 Jan '09

21 Jan '09

Hi, I have Installed Freeswitch in Redhat5.1 , but when i start the freeswitch for /usr/local/freeswitch/bin throws the following error 2009-01-21 13:54:01 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2009-01-21 13:54:01 [ERR] sofia.c:1815 config_sofia() Failed to get external ip. i am able to ping stun.freeswitch.org but not allowing when specified stun.freeswitch.org:3478 , the linux doesn't have any firewall and connected with internet please help me , do i need to configure any thing else ? Regards, Sanju. _____________________________________________________________ Gift Certificates http://www.online-gift-certificate.com This email was sent using 11hit.com free web-based email! http://www.11hit.com

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Is there any plans to implement some kind of measures and statistics function?
by seven du 21 Jan '09

21 Jan '09

Hi developers, Now and then we need to know some statistics message and sure we need some way to measure it. My ideas just like this: http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Exte… Is there some plans to build this kind of function in? If not, can some one tell me how easy or hard to do it? I think it would be better to use pcap or 3rd party stuffs the do that. Thanks. Seven

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Latest build - Socket inbound variable_* on custom event problem.
by Johny Kadarisman 20 Jan '09

20 Jan '09

Hi all, Is there a way to propagate variable_* into custom event. I'm connecting with inbound socket, and listening to that custom events. during recent trunk update, I encountered an issues from external application, after look around, found out that these value is not in the events. After search around, found out some reference about "verbose_events" with outbound sockets. So, I insert that applications in dialplan, but still don't see any variable_* being propagate. Is there anyway to enable this for inbound sockets? Thanks, Johny K.

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Is it possible to add a new profile configuration and enable it without restarting the server?
by seven du 20 Jan '09

20 Jan '09

Hi, For production use, we want the service run consistently and long as possible. As we have command tools to reload xml configurations and restart gateways, I have a problem to make my configuration take effect before restarting the whole FreeSWITCH server. Here are two questions: 1) When I add a gateway to a profile, say external, I can do reloadxml and sofia profile external restart. But if I add a new profile, how can I do it as it will complain the profile name not exist? 2) Is it possible to enable/disable/restart a single gateway? Thanks Seven

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Re: [Freeswitch-dev] mod_skypiax inching forward
by seven du 19 Jan '09

19 Jan '09

Giovanni, Thank you for taking mu suggestions take into account. Another idea: Make the skype in context and dialplan configurable(other than 5000?) maybe useful. But not sure, due to the limits of skype in, perhaps the only use of skype in is transfer into an ivr. And, Just found a easy way to create bunch of skype configurations. If you have a bunch of skype accounts like skypiax1, skypiax2, skypiax3 ... You only have to login into skypiax1 one time and chose the right sound device and other settings like disable events etc. and you will get a dir in {your home dir}/.Skype/skypiax1. Just copy skypiax1 into skypiax2, skypiax3... They will have similar configurations, as long as they share a same password. Cheers- Seven > Seven, > thanks for suggestions, I'll start on them asap! > > Do you have other ideas/suggestions/hints on how to make skypiax more useful/easy to use?

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