- FreeSWITCH-users - FreeSWITCH Mailing Lists

Notify of a missed call (no voicemail left)
by Ryan How 22 Feb '12

22 Feb '12

Hi, Is there any way to send an email notification of a missed call, no voicemail left?. Or even if a voicemail is left I guess, that isn't a biggy... just so if someone rings and hangs up, or rings and voicemail answers but no message. I'm thinking something that goes in here like: <action application="email" data="something here"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <insert in here> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> But that wouldn't capture when they hang up before voicemail answering would it ? I guess I could do something like application="socket" and then write an event socket program that sends an email. If there isn't already something out there that does it. Otherwise I'll leave this for another day coz it is a bit beyond my current magic ability, I need to level up :P Thanks again! Ryan

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Park an incoming call without answering it?
by Chris Martineau 22 Feb '12

22 Feb '12

Hi, Is it possible to somehow park or just make the call wait on the system without bridging it or answering it. I would like the outside party to still be in ringing state so I can the pick it up. Need something like a virtual ringing destination but cannot see how to implement? Any ideas Many thanks Chris

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Modify INVITE format
by liu rush 22 Feb '12

22 Feb '12

Dear all, Does anyone know how to modify the INVITE format of FS? Regard, Rush

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multi-FS setup and sip presence
by Vik Killa 22 Feb '12

22 Feb '12

So it appears i'm not the only person having issues with presence in a multi-FS setup... http://jira.freeswitch.org/browse/FS-3758 Currently presence is only works when a call comes to a UA on the FS server that UA is registered to... It looks like this patch was submitted to GIT but there is no clear instruction on how to implement it. Should a sofia parameter be used like <param name="shared-subscriptions" value="true"/> ? Also, I've been told i need to set this parameter: <param name="dbname" value="shared_presence"/> My question is what should the value of that parameter be? or does that not matter because FS automatically creates the SCHEMA? All my FS systems are connected to the same ODBC (in sofia.conf.xml and switch.conf.xml) And each FS system has a unique hostname. I'm told that if I set the hostname to the same on all FS boxes, presence will work, but at the same time I'm told setting the hostname of all FS boxes to the same is not good practice in a clustered setup. Also, I have the presence_id setup in the directory like <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> Any help would be much appreciated, thanks.

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(no subject)
by hjz51 22 Feb '12

22 Feb '12

Who have mod_sofia.c and switch_rtp.c source’s development documents. Can you give me to look at, thanks！

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mod_mongo usage
by ovvenkat 22 Feb '12

22 Feb '12

Hi to all, Any one help me to connect mongodb through mod_mongo module. thanks in advance. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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Where to best get help with freeswitch on gentoo?
by Marcus Mülbüsch 22 Feb '12

22 Feb '12

Hello all, I make my first steps with freeswitch, and following the guide from http://wiki.freeswitch.org/wiki/Installation:Gentoo I try to get freeswitch running. However, depending on the USE-Flags set, or the FREESWITCH_MODULES used the compile fails; but I cannot find out why. For example, when I try the module "shout", compile fails with * ERROR: net-misc/freeswitch-9999 failed (compile phase): * failed to build FreeSWITCH * * Call stack: * ebuild.sh, line 85: Called src_compile * environment, line 6473: Called die * The specific snippet of code: * emake MONO_SHARED_DIR="${T}" || die "failed to build FreeSWITCH"; Googling around did not really lead me to anything useful. Is gentoo using the freeswitch ebuild still a (more or less) recommended way to run freeswitch? Can I ask more specific questions here, or somewhere else? Marcus

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Hardware SIP device register problem (Planet VIP-161W)
by Oleg Belykh 22 Feb '12

22 Feb '12

The device can't get registered on freeswitch system with "481 Call Does Not Exist" connection log: recv 417 bytes from tcp/[172.16.20.20]:2055 at 16:38:37.513380: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com SIP/2.0 Via: SIP/2.0/TCP 172.16.20.20:5060;rport;branch=z9hG4bK57a5b7e300000 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com> Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332939 REGISTER Max-Forwards: 70 Contact: <sip:1105@172.16.20.20:5060;transport=tcp>;expires=900;events="message-summary" User-Agent: VOIP_Agent_001 Supported: path Content-Length: 0 ------------------------------------------------------------------------ send 623 bytes to tcp/[172.16.20.20]:2055 at 16:38:37.521226: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 172.16.20.20:5060;rport=2055;branch=z9hG4bK57a5b7e300000 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com>;tag=vy75Dp57NUm8m Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332939 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="kustogroup.com", nonce="20781b2c-185b-e111-952d-002590016844", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 657 bytes from tcp/[172.16.20.20]:2055 at 16:38:37.538670: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com SIP/2.0 Via: SIP/2.0/TCP 172.16.20.20:5060;rport;branch=z9hG4bKc91f046100001 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com>;tag=vy75Dp57NUm8m Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332940 REGISTER Max-Forwards: 70 Contact: <sip:1105@172.16.20.20:5060;transport=tcp>;expires=900;events="message-summary" User-Agent: VOIP_Agent_001 Supported: path Authorization: Digest username="1105", realm="kustogroup.com", nonce="20781b2c-185b-e111-952d-002590016844", qop=auth, cnonce="53506aaa", nc=00000001, uri="sip:kustogroup.com", response="26cb514fc66213aef7873b857a5a42a6" Content-Length: 0 ------------------------------------------------------------------------ send 508 bytes to tcp/[172.16.20.20]:2055 at 16:38:37.540118: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/TCP 172.16.20.20:5060;rport=2055;branch=z9hG4bKc91f046100001 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com>;tag=vy75Dp57NUm8m Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332940 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ Please help, i cant understand the problem, another hardware device (DLink gateway) got registered without any problem. Connection logs are similar. ------------------------------------------------------------------------ recv 406 bytes from udp/[172.16.0.25]:5060 at 16:35:54.213419: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe8a46c80fd1882b6 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com> Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4722 REGISTER Contact: <sip:1999@172.16.0.25:5060> Expires:600 Max-Forwards:70 User-Agent:dlink 12-38-43929367-0.9.5.1.1928-SA7O4 Content-Length: 0 ------------------------------------------------------------------------ send 635 bytes to udp/[172.16.0.25]:5060 at 16:35:54.220591: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe8a46c80fd1882b6 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com>;tag=p8FH3K0j75erF Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4722 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="kustogroup.com", nonce="dcc9c5ca-175b-e111-952d-002590016844", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 639 bytes from udp/[172.16.0.25]:5060 at 16:35:54.311703: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe489a6a305c49f70 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com> Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4723 REGISTER Contact: <sip:1999@172.16.0.25:5060> Expires:600 Max-Forwards:70 Authorization:Digest username="1999",realm="kustogroup.com",nonce="dcc9c5ca-175b-e111-952d-002590016844",uri="sip:kustogroup.com:5060",response="2ea94376953abc90a78efaa831e79af4",algorithm=MD5,cnonce="3870847b",qop=auth,nc=00001273 User-Agent:dlink 12-38-43929367-0.9.5.1.1928-SA7O4 Content-Length: 0 ------------------------------------------------------------------------ send 590 bytes to udp/[172.16.0.25]:5060 at 16:35:54.341527: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe489a6a305c49f70 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com>;tag=QH994eHp4e5aB Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4723 REGISTER Contact: <sip:1999@172.16.0.25:5060>;expires=600 Date: Sun, 19 Feb 2012 16:35:54 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0

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FreeSWITCH Conference Call Tomorrow (Wed Feb 22)
by Michael Collins 21 Feb '12

21 Feb '12

Hey all! Yes, I promise that I will come up for air and hang out with the community tomorrow. :) We have a special guest joining us tomorrow: Stefan Wintermeyer from Amooma. He will be talking about Gemeinschaft v4 and all the interesting things that lead to going with FreeSWITCH instead of Asterisk. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_22 Look forward to speaking with you all tomorrow! -Michael

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SIP bridge
by Anto 21 Feb '12

21 Feb '12

Hello I have a internal equipment and I want to use a gateway FreeSWITCH servers against some providers. Let me know if it is possible to transfer SIP messages (bridge), such as early media, etc.. I would like these messages to stay at the gateway and not reach the supplier, Is possible?. I've been looking for information but have not found anything about it :-(. Thank you very much. Best regards Anto

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